irclog2html for #asterisk on 20070125

`00:00.02`	`JT`	`99.97% or higher is fine`
`00:00.11`	`mercestes`	`not at idle. You dump processes on there he's going to drop down to 98-97 and that sucks.`
`00:00.11`	`JT`	`mercestes: so long as it doesn't go down under load`
`00:00.16`	`mercestes`	`yea, if it stays at 99.97.`
`00:00.24`	`JT`	`zttest is a lot of voodoo more than anything`
`00:00.48`	`JT`	`mercestes: not so, i have systems running constant 99.97 or 99.98 at idle that run the same during load and work fine`
`00:00.50`	`mercestes`	`Probably. I'd give it a whirl under some test load then to be sure.`
`00:00.54`	`HushPe`	`like i said it's not sounding crackly any more (if any it's bugger all, probably just a gain thing)`
`00:01.13`	`mercestes`	`I run 100% straight through, even with call volume`
`00:01.21`	`JT`	`HushPe: try running some calls, and some load on the system, see if it gets the shits`
`00:01.23`	`mercestes`	`light call volume`
`00:01.29`	`SomeOne1`	`JT: jesus christ thats a lot of bandwidth`
`00:01.35`	`JT`	`mercestes: you need magic hardware to do that :)`
`00:01.46`	`HushPe`	`mercestes: not with a 102NZD motherboard`
`00:01.46`	`JT`	`SomeOne1: no shit, 2000 calls is a lot`
`00:01.48`	`HushPe`	`with 2 pci slots <g>`
`00:02.04`	`JT`	`mercestes: what is your hardware?`
`00:02.19`	`mercestes`	`PE something or another.`
`00:02.24`	`mercestes`	`Intel Xeon 64bit`
`00:02.47`	`clorabit`	`JT: i've try connect without password asterisk log show message restricting registration for peer '1234' to 60 seconds is that ok ?`
`00:02.48`	`mercestes`	`4g memory. SCSI drives.`
`00:02.51`	`mercestes`	`you know, the standard.`
`00:02.59`	`JT`	`mercestes: yeah, zttest has always favoured dell from what i hear`
`00:03.07`	`JT`	`mercestes: dual proc?`
`00:03.15`	`mercestes`	`yea`
`00:03.20`	`mercestes`	`dual core rather.`
`00:03.23`	`mercestes`	`same diff.`
`00:03.39`	`mercestes`	`yea, most of my zttests have been on dells so you may have me there.`
`00:03.44`	`JT`	`clorabit: you don't need to register, with no auth, not sure if you can get incoming calls on the softphone though`
`00:04.03`	`JT`	`it was probably designed around a dell`
`00:04.23`	`JT`	`mercestes: HushPe's worst score before was 72%, this is a MASSIVE improvement`
`00:04.59`	`mercestes`	`True that.`
`00:05.06`	`SomeOne1`	`JT: for 2000 calls, i calculated: 95.2kbits/sec * 2-way call = 190.4kbits/sec = 0.1859375mbits/sec * 2000 = 371.875`
`00:05.08`	`SomeOne1`	`mbps`
`00:05.23`	`JT`	`it's 85kbit/s`
`00:05.24`	`mercestes`	`that's only what, 200 t1's?`
`00:05.31`	`data23`	`I thought ulaw was 64? :)`
`00:05.37`	`SomeOne1`	`with all headers`
`00:05.39`	`SomeOne1`	`and ethernet`
`00:05.41`	`SomeOne1`	`overhead`
`00:05.42`	`SomeOne1`	`and stuff`
`00:05.44`	`data23`	`ah yes`
`00:05.50`	`SomeOne1`	`it comes out to be 95.2 doesnt it?`
`00:05.56`	`SomeOne1`	`http://www.bandcalc.com/`
`00:05.58`	`HushPe`	`JT: 2 calls outgoing 73 passes: Best: 99.987793 -- Worst: 99.975586 -- Average: 99.975753`
`00:05.59`	`JT`	`well all sip overhead brings it to 85kbit/s`
`00:06.07`	`HushPe`	`so it looks like it's holding`
`00:06.11`	`JT`	`no idea about ethernet overhead`
`00:06.18`	`SomeOne1`	`but its not always true that in a normal conversation both people are talking`
`00:06.24`	`SomeOne1`	`and if theres silence suppression`
`00:06.27`	`SomeOne1`	`that can be cut in half`
`00:06.35`	`JT`	`i hate silence supression, it sounds awful`
`00:06.42`	`data23`	`indeed`
`00:06.52`	`SomeOne1`	`well without silence suppression BOTH sides will be transmitting continusously?`
`00:06.58`	`JT`	`because it's always implemented shit`
`00:06.59`	`JT`	`yes`
`00:07.03`	`SomeOne1`	`or will one side stop transmitting when the other side is talking?`
`00:07.04`	`clorabit`	`JT: i try to dial echo extension asterisk log show that it spawn extensions but nothing happen in client no sound at all`
`00:07.23`	`JT`	`SomeOne1: both will transmit, like a normal phone line`
`00:07.27`	`SomeOne1`	`god damnit`
`00:07.31`	`SomeOne1`	`thats so much bandwidth`
`00:07.36`	`SomeOne1`	`do you have any idea how much that will cost`
`00:07.37`	`bmd`	`somebody needs to write an adaptive comfort noise generator that listens to the "silence" that it's suppressing`
`00:07.39`	`SomeOne1`	`jeesus`
`00:07.41`	`JT`	`SomeOne1: so, where's the bandwidth going?`
`00:07.45`	`JT`	`SomeOne1: Internet?`
`00:07.51`	`SomeOne1`	`yeah`
`00:07.53`	`data23`	`heh`
`00:07.58`	`mercestes`	`Goodluck`
`00:08.01`	`data23`	`hope you have a big cheque book`
`00:08.09`	`JT`	`bmd: gsm silence supression is much better than sip rtp, it detects it better, and inserts whitenoise`
`00:08.14`	`mercestes`	`I hope you have your own WAN.`
`00:08.21`	`SomeOne1`	`well`
`00:08.25`	`SomeOne1`	`how the hell does level 3 do it?`
`00:08.25`	`JT`	`sip rtp silence supression is dumb dumb dumb`
`00:08.30`	`HushPe`	`JT: next problem is my incoming trunk not being picked up correctly asterisk detects the ring, but it's not actually opening up the line`
`00:08.30`	`SomeOne1`	`im sure they do like`
`00:08.38`	`data23`	`SomeOne1: they have big fat junipers`
`00:08.39`	`SomeOne1`	`a million calls concurrently`
`00:08.40`	`mercestes`	`Level 3 has their own infrastructure`
`00:08.43`	`data23`	`SomeOne1: and their own network`
`00:08.43`	`HushPe`	`but outgoing calls via the same card(s) work great`
`00:08.46`	`JT`	`SomeOne1: how come you need to handle 2000 calls?`
`00:08.58`	`mercestes`	`Time Warner runs over their net for their cable service in some areas.`
`00:09.00`	`JT`	`SomeOne1: probably have a real phone exchange that uses TDM, not voice`
`00:09.01`	`SomeOne1`	`putting up a PBX`
`00:09.02`	`mercestes`	`They ARE the backbone.`
`00:09.05`	`JT`	`but even voice is doable`
`00:09.17`	`JT`	`SomeOne1: you do have 2000 calls to put on it?`
`00:09.20`	`SomeOne1`	`yep`
`00:09.25`	`SomeOne1`	`i really do`
`00:09.25`	`SomeOne1`	`:)`
`00:09.26`	`JT`	`where from`
`00:09.28`	`JT`	`well`
`00:09.30`	`SomeOne1`	`pakistan`
`00:09.34`	`JT`	`i assume you have an income stream`
`00:09.36`	`SomeOne1`	`and india`
`00:09.39`	`JT`	`from those calls`
`00:09.41`	`SomeOne1`	`yes`
`00:09.42`	`mercestes`	`SomeOne1: Call centers?`
`00:09.43`	`JT`	`if not, there's an issue`
`00:09.52`	`JT`	`so then you can afford it`
`00:09.55`	`SomeOne1`	`what do you mean income stream? like.. monetary?`
`00:09.59`	`SomeOne1`	`of course an income stream`
`00:10.00`	`JT`	`you don't have to run g.711 though`
`00:10.01`	`data23`	`surely if you're just on about doing least cost routing, you'll want them locally in each country anyway`
`00:10.02`	`JT`	`yes`
`00:10.34`	`SomeOne1`	`well if you wanted to give a company in the USA termination in india over SIP`
`00:10.40`	`SomeOne1`	`or well, anyway else`
`00:10.45`	`SomeOne1`	`how the hell else could you do it`
`00:10.49`	`SomeOne1`	`besides over internet`
`00:10.52`	`JT`	`SomeOne1: have you checked indian law?`
`00:10.58`	`data23`	`you'll probably find 300mbits is the total bandwidth for india =)`
`00:11.04`	`SomeOne1`	`JT: dont worru about that :)`
`00:11.15`	`JT`	`SomeOne1: there's been a big crackdown on voip there`
`00:11.22`	`SomeOne1`	`JT: explain`
`00:11.27`	`SomeOne1`	`what about DIDs?`
`00:11.38`	`JT`	`err they have a govt run telco iirc`
`00:11.49`	`mercestes`	`as in their infrastructure blocks it if they can identify what it is and they come chasing after you.`
`00:11.49`	`JT`	`and it's a monopoly`
`00:12.06`	`SomeOne1`	`recently theyve lifted this ban though`
`00:12.07`	`mercestes`	`iirc they hot spot data encryption as well.`
`00:12.08`	`data23`	`aye, even the likes of BT have trouble with places like that`
`00:12.11`	`JT`	`voip is only allowed for businesses purposes like callcentres`
`00:12.13`	`SomeOne1`	`okay forget india for a second, i know pakistan does it`
`00:12.22`	`JT`	`i see`
`00:12.27`	`SomeOne1`	`still`
`00:12.35`	`mercestes`	`SomeOne1: Well, first, I wouldn't run SIP over the ocean to pakistan.`
`00:12.39`	`SomeOne1`	`how would you transport like 100,000 minutes between india and pakistan`
`00:12.47`	`SomeOne1`	`okay what would you use`
`00:12.51`	`mercestes`	`I would establish my own honkin' data channel from the US to Pakistan and dump a PBX in pakistan`
`00:12.56`	`JT`	`do you actually have 2000 Erlangs of traffic, or are you just hoping you'll have that much?`
`00:12.57`	`mercestes`	`and I would IAX that bitch from box to box.`
`00:13.20`	`SomeOne1`	`hahaha`
`00:13.26`	`data23`	`SomeOne1: so let me get this straight, you have someone in pakistan that has a E1 channel bank, that will convert the signals from their phones to SIP and wants to deliver it to you in the USA? You then want to 'route it' for them elsewhere? i.e. India or whatnot?`
`00:13.26`	`SomeOne1`	`honkin data channel?`
`00:13.51`	`SomeOne1`	`yes i want them to deliver it here`
`00:13.52`	`SomeOne1`	`to the USA`
`00:13.56`	`mercestes`	`what do you call a dedicated 350 mbps of throughput from the US to Pakistan?`
`00:14.05`	`SomeOne1`	`non-existent`
`00:14.13`	`mercestes`	`good call`
`00:14.15`	`SomeOne1`	`dude how does like at&t do it`
`00:14.19`	`SomeOne1`	`then`
`00:14.21`	`SomeOne1`	`explain to me`
`00:14.26`	`SomeOne1`	`people call pakistan`
`00:14.27`	`JT`	`they have money`
`00:14.27`	`SomeOne1`	`and india`
`00:14.33`	`JT`	`quite easy then`
`00:14.36`	`mercestes`	`AT&T owns the lines you are trying to lease.`
`00:14.42`	`JT`	`SomeOne1: you talking about the normal phone network?`
`00:14.48`	`SomeOne1`	`yeah`
`00:14.49`	`data23`	`the likes of AT&T have their big atm network around the world`
`00:14.54`	`SomeOne1`	`what is an E1 channel?`
`00:14.56`	`data23`	`cables like TAT14 etc`
`00:14.57`	`JT`	`SomeOne1: international TDM network`
`00:15.04`	`data23`	`E1 = PRI (euro version of your T1)`
`00:15.05`	`JT`	`SomeOne1: seriously...... learn more about telcoms`
`00:15.36`	`JT`	`there are massive undersea fibre cable networks that run TDM voice`
`00:15.39`	`data23`	`SomeOne1: http://en.wikipedia.org/wiki/Transatlantic_telephone_cable`
`00:15.41`	`JT`	`they don't use voip usually`
`00:15.42`	`data23`	`^ go read that`
`00:15.50`	`SomeOne1`	`okay call`
`00:15.54`	`SomeOne1`	`screw voip`
`00:15.56`	`SomeOne1`	`im in the wrong channel`
`00:15.58`	`SomeOne1`	`heh`
`00:16.00`	`JT`	`so they run 64kbit/s each way per call`
`00:16.00`	`SomeOne1`	`cool*`
`00:16.04`	`SomeOne1`	`if anyone wants it`
`00:16.18`	`SomeOne1`	`i can get you the cheapest termination rate in india and pakistan`
`00:16.52`	`mercestes`	`Um....Asterisk isn't just voip, ya know.`
`00:16.58`	`JT`	`SomeOne1: the way most cheap voip providers do it is by running highly compressed codecs`
`00:17.03`	`JT`	`that save bandwidth`
`00:17.08`	`JT`	`at the expense of call quality`
`00:17.09`	`data23`	`and sound like tin cans :)`
`00:17.34`	`data23`	`hence why most calls to those sort of destinations, always sound tinny :)`
`00:17.42`	`SomeOne1`	`hmmm`
`00:17.46`	`SomeOne1`	`so we need to lay a PTAT-1`
`00:18.10`	`data23`	`if you want to go to Ireland via Bermuda, sure :)`
`00:18.15`	`SomeOne1`	`lol`
`00:18.22`	`data23`	`and you have a spare $400 million`
`00:18.26`	`SomeOne1`	`you all are mocking me, but at good reason`
`00:18.34`	`SomeOne1`	`i dont know about this stuff`
`00:18.37`	`clorabit`	`hello do i need sound card to run asterisk ??`
`00:18.44`	`[TK]D-Fender`	`clorabit : nope`
`00:18.44`	`data23`	`clorabit: no :)`
`00:19.44`	`clorabit`	`when i try to dial my extension using command dial it show error unable to re-open DSP device /dev/dsp what this mean ?`
`00:19.44`	`SomeOne1`	`what is erlangs?`
`00:20.12`	`HushPe`	`JT: i have a GSM connected to one incoming line, that works perfectly except i can't dial extensions for some reason... this pastie: http://pastie.caboo.se/35453 << real land line, it rings, asterisk doesn't seem to open the line up (SIP/Zap phone rings). If the user hangs up I get a 'call back' with a dial tone.`
`00:21.50`	`JT`	`SomeOne1: a telecommunications traffic engineering term`
`00:21.50`	`JT`	`1 Erlang = 1 circuit in use for 1 hour`
`00:21.51`	`SomeOne1`	`okay so to transport 500,000 minutes per month transcontinent... VoIP is not a good idea`
`00:21.51`	`JT`	`it is, if you have the bandwidth`
`00:21.51`	`bkw_`	`TOP THAT BABY`
`00:21.53`	`JT`	`and resources in general`
`00:21.58`	`data23`	`whatever you do, if you don't have the infrastructure, it's gonna cost big time`
`00:22.06`	`bkw_`	`you can now use your Asterisk speech recognition ports with FreeSWITCH`
`00:22.17`	`JT`	`voip over Internet is less reliable than TDM though`
`00:22.24`	`SomeOne1`	`TDM`
`00:22.35`	`JT`	`time division multiplexing`
`00:22.48`	`anthonyl`	`so what is this asterisk thing?`
`00:22.51`	`SomeOne1`	`ahh`
`00:22.53`	`anthonyl`	`can i make calls with it?`
`00:23.03`	`bkw_`	`anthonyl, did you see what we just checked in?`
`00:23.05`	`anthonyl`	`like over the regular phone ports?`
`00:23.15`	`JT`	`this pastie.caboo.se site is quite slow`
`00:23.16`	`anthonyl`	`ssh im being funny!`
`00:23.26`	`anthonyl`	`bkw_, should i check the fisheye?`
`00:23.34`	`bkw_`	`yes`
`00:23.36`	`bkw_`	`please do`
`00:24.02`	`anthonyl`	`<PROTECTED>`
`00:24.03`	`anthonyl`	`word`
`00:25.13`	`bkw_`	`anthonyl, so boi what ya think?`
`00:25.18`	`bkw_`	`our ASR connector is open source`
`00:26.05`	`anthonyl`	`promising`
`00:26.38`	`*** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no)`
`00:26.56`	`*** join/#asterisk jm\|laptop (n=jm\|home@dilbert.jamiem.com)`
`00:27.03`	`*** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com)`
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`00:28.17`	`data23`	`right bedtime, 00:39. nn all`
`00:28.36`	`wunderkin`	`sync your clock`
`00:28.43`	`data23`	`i was just thinking that :)`
`00:29.02`	`Mad\|Cow`	`Hi eveyone, I'm trying to set option 150 in my dchpd.conf on a Ubuntu box. Does anyone have an example I could copy from?`
`00:29.20`	`JT`	`what does that have to do with asterisk`
`00:29.24`	`data23`	`consider it synced (00:29 :p) <- gone`
`00:29.46`	`Mad\|Cow`	`jt: option 150 is what the cisco phones use to TFTP boot from`
`00:30.25`	`JT`	`right, well i'm sure dhcpd.conf is well documented`
`00:30.35`	`JT`	`i'm not sure what the exact parameter is`
`00:30.50`	`Mad\|Cow`	`JT: I wish it were true... but its not in there`
`00:31.35`	`JT`	`http://www.google.com.au/search?hl=en&ie=ISO-8859-1&q=dhcpd.conf+option+150&meta=`
`00:32.23`	`SomeOne1`	`JT: can you lease a dedicated TDM?`
`00:32.27`	`SomeOne1`	`between pakistan and here?`
`00:32.32`	`Mad\|Cow`	`JT: yeah... I have google'd it a couple of times... most of the references seem to be on RedHat. Doesnt work with Ubuntu's version of dhcpd`
`00:32.40`	`*** join/#asterisk Zodiacal (n=hehehe@bdsl.66.14.242.199.gte.net)`
`00:33.09`	`Zodiacal`	`anyone know how i could plug in a voip phone at home and have it connect to my asterisk server at work via my vpn?`
`00:33.20`	`Zodiacal`	`considering the phone doesn't have a vpn client :P`
`00:33.49`	`Zodiacal`	`i guess i could add a nic to my home pc and use ICS`
`00:34.01`	`Zodiacal`	`or would that even be on the same network?`
`00:34.05`	`Zodiacal`	`proably not`
`00:34.07`	`Zodiacal`	`hrmm..`
`00:35.13`	`rudholm`	`can you run your vpn client on your router?`
`00:35.28`	`rudholm`	`that way anything on your LAN can see the vpn`
`00:36.28`	`JT`	`SomeOne1: yes, if one is available`
`00:36.34`	`SomeOne1`	`how much will it cost?`
`00:37.36`	`JT`	`i don't know, those sort of things are not advertised in price`
`00:37.39`	`JT`	`you may not need one`
`00:37.43`	`JT`	`depends what you're doing`
`00:38.09`	`SomeOne1`	`i'm transporting 2000 concurrent calls`
`00:40.17`	`*** join/#asterisk Zaw (i=zaw@unaffiliated/zaw)`
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`00:45.23`	`litage`	`is there any particular reason why asterisk doesn't have a log message priority of "info"?`
`00:46.35`	`james_`	`because what it does is secret`
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`00:55.07`	`JunK-Y`	`SomeOne1: on 1 single box?`
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`01:28.22`	`[hC]`	`anyone know if any of polycoms video conferencing products do sip w/ asterisk?`
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`01:48.56`	`test34`	`Is there anyway to unlock Vonage's Motorola VT2442 that has recently been connected to the internet (ie today)?`
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`01:55.10`	`demigod2k`	`hi`
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`01:59.13`	`HushPe`	`is there a simple reference that lets me connect a SIP line as an incoming/outgoing trunk line?`
`01:59.39`	`HushPe`	`so i can 'dial 9' then my number and it uses that trunk`
`02:02.37`	`*** join/#asterisk bricecubed (n=nesta@pool-72-84-202-204.rcmdva.east.verizon.net)`
`02:06.12`	`bricecubed`	`I am new to the PBX world && have a (likely idiotic) question; I would like to deploy Asterisk as a "gateway" for the telephones to use in the house. I would also like the ability to put calls on hold and intercom the family letting them know so & so is on hold ;) . The telephones are currently hooked up to a POTS line.. and I wired (internally) in daisy chain fashion. My ? is must I re-wire? e.g. run a seperate wire to ea`
`02:06.12`	`bricecubed`	`ch room in order to have hold/intercom functionality? Also, is Asterisk capable of this?`
`02:07.04`	`demigod2k`	`bricecubed: the most common case is that you're running CAT5 ethernet in a star fashion back to a central ethernet switch, out to VOIP phones.... so yes`
`02:07.36`	`*** join/#asterisk infernix (n=nix@spirit.infernix.net)`
`02:10.22`	`bricecubed`	`demigod2k, OK.. do the VOIP phones support intercom, etc? I remember doing this once using the 4 wire phone line going to a panasonic box`
`02:10.46`	`bricecubed`	`so there's no special phones, etc. that would allow me to do this on the existing wiring?`
`02:11.54`	`demigod2k`	`bricecubed: it depends on the phone and I haven't configured the polycoms at work for it yet... but it is "possible" to configure for speakerphone (2way) or paging (1way)`
`02:12.11`	`bricecubed`	`demigod2k, I see`
`02:12.11`	`demigod2k`	`all you mean by intercom is "autoanswer speakerphone" I guess`
`02:12.32`	`bricecubed`	`yeah.. a page that says.. "pikcup the phone!" so people know to pickup ;)`
`02:13.03`	`bricecubed`	`hmm.. is registration maxed out/disabled @ asterisk.org forums?`
`02:13.08`	`demigod2k`	`basically. I know its possible (maybe not on everything) to do an autoanswer paging type thing`
`02:13.12`	`*** join/#asterisk j0 (n=dan@S01060016b6b541d2.va.shawcable.net)`
`02:15.08`	`demigod2k`	`bricecubed: I mean... not really. It's called POTS service.`
`02:15.24`	`demigod2k`	`ethernet goes back to a switch/hub, so you can't really daisychain`
`02:15.37`	`demigod2k`	`plus you need higher quality twisted-pair wire for ethernet. its just not really an option`
`02:16.28`	`*** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-130.mtco.com)`
`02:16.55`	`bricecubed`	`ahh OK`
`02:17.27`	`bricecubed`	`I thought my asterix box would be full of those digium cards that can handle the RJ11 connection`
`02:17.28`	`demigod2k`	`was this a recently built house? odd to wire phone like that when copper wire is just so cheap (except recently)`
`02:17.50`	`bricecubed`	`but.. I guess that's more expensive than running cat5 & using VOIP phones?`
`02:17.50`	`demigod2k`	`well sure. 1 RJ-11 = 1 extension. just like your old POTS service`
`02:18.00`	`bricecubed`	`built 1966`
`02:18.18`	`demigod2k`	`so then you have totally fucked up service where everything on one daisy-chain is the same extension (which you surely dont want)`
`02:18.20`	`bricecubed`	`right.. so it sounds like I may as well run cat5`
`02:18.30`	`bricecubed`	`demigod2k, right`
`02:18.38`	`demigod2k`	`totally. if you're in a ranch it cant be that tough either`
`02:19.45`	`demigod2k`	`even in a 2-level just run one bundle of cables up to your attic and then down from there`
`02:20.04`	`demigod2k`	`when I "rewired" my house it was little more than a 1 week task, with a beer in hand every night`
`02:21.00`	`bricecubed`	`:) nice`
`02:21.38`	`bricecubed`	`you come to Richmond, we'll give you 2 beers every night ;)`
`02:21.41`	`*** join/#asterisk stkn__ (i=nobody@gentoo/developer/pdpc.active.stkn)`
`02:22.01`	`Supaplex`	`I'm considering fiber+media converters for each room (or sx adapter).`
`02:23.25`	`*** join/#asterisk angom (n=angom@red-corp-201.143.146.171.telnor.net)`
`02:23.48`	`JT`	`anyone know where the ISDN Subaddress is in pri intense debug, or alternatively, how to get it from the dialplan?`
`02:25.15`	`demigod2k`	`nope sorry`
`02:25.25`	`demigod2k`	`isdn = it still does nothing`
`02:26.11`	`JT`	`demigod2k: err, what do you use?`
`02:27.42`	`demigod2k`	`actually my family had ISDN for years and just switched off finally to DSL`
`02:27.57`	`demigod2k`	`we use a digium with POTS on the new asterisk system at work (just hang out here for fun, I kinda do it on the side)`
`02:28.04`	`JT`	`i'm talking about voice, not data`
`02:28.10`	`demigod2k`	`ISDN is awesome the rates are just very, very, very unfavorable around Michigan`
`02:28.17`	`JT`	`pots is far inferior to isdn`
`02:28.19`	`JT`	`heh`
`02:28.45`	`demigod2k`	`Ameritech then SBC then AT&T. Nobody has charged a reasonable amount of money for it. I think they decided that only ATMs need it so they should charge like crazy for it`
`02:29.07`	`xpot`	`anyone know of a way to write data to db on hangup? So if the caller calls in and hangs up the call after a few seconds (or anytime during call) write to db`
`02:29.15`	`JT`	`yeah i hear that's the story in north america`
`02:29.19`	`JT`	`a PRI is also ISDN, btw`
`02:29.22`	`J4k3`	`here in texas we just didn't get ISDN til it was very much out of style`
`02:29.37`	`demigod2k`	`ya NA it's insane...... hence the acronym`
`02:29.37`	`J4k3`	`but, at least it was flat rate`
`02:29.38`	`rudholm`	`I'd like to switch from POTS to ISDN at home but a) AT&T (nee "SBC" nee "Pacific Bell") doesn't offer CNAM on BRI and Digium's BRI card is for EuroISDN not National ISDN.`
`02:29.42`	`demigod2k`	`my boss used to work in telecom back at Mot and had to laugh. its awful here`
`02:29.51`	`demigod2k`	`its effectively dead - DSL took over as a profit center`
`02:29.52`	`JT`	`CNAM?`
`02:29.58`	`rudholm`	`Calling Name`
`02:30.10`	`JT`	`err are you sure digium's card doesn't work on national isdn`
`02:30.18`	`demigod2k`	`we paid per-minute for data/voice on ISDN -- absolutely no way to get an "unlimited" plan despite it being digital`
`02:30.25`	`JT`	`digium aren't the only people who make isdn cards either`
`02:30.40`	`Supaplex`	`but digium invented ISDN ;)`
`02:30.48`	`Supaplex`	`nawt! :P`
`02:31.12`	`rudholm`	`yeah, I'm pretty sure it's Euro only`
`02:31.24`	`J4k3`	`oh well... ISDN BRI = 24 hour repair time... my data T1 has 4 hour max repair time... and the voip provider is "24 hour onsite with best effort response time"... Beats the hell out of the "oh no its the weekend" situation where the line dies friday night and its promised to be fixed by monday evening.`
`02:32.07`	`rudholm`	`digium isn't the only BRI card maker, but I don't know of any other BRI card that is known to work with Asterisk. Do you?`
`02:32.18`	`JT`	`rudholm: i really doubt that`
`02:32.33`	`JT`	`lol there are tonnes of BRI cards that have worked with asterisk for years`
`02:32.49`	`Strom_C`	`but I doubt any of them support NI2`
`02:33.16`	`JT`	`any evidence, or just a feeling you have? :)`
`02:33.33`	`Strom_C`	`considering they're all for the european market....just a feeling`
`02:33.39`	`rudholm`	`the evidence is that I haven't been able to find one.`
`02:33.46`	`rudholm`	`that's why I asked if you knew of one. I've been asking everyone.`
`02:34.03`	`JT`	`i thought it was more a software issue`
`02:34.22`	`rudholm`	`either way, lack of support is lack of support.`
`02:34.22`	`JT`	`and if you're using software like bristuff that uses zaptel, you can choose national isdn`
`02:34.23`	`*** join/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum)`
`02:34.32`	`JT`	`assumed lack of support`
`02:34.37`	`JT`	`documentation may be poor`
`02:34.55`	`Cherebrum`	`It seems someone has been editing the Asterisk_PBX wikipedia entry: http://tinyurl.com/3xcjb3`
`02:36.55`	`rudholm`	`that criticism sounds like someone has an axe to grind.`
`02:37.38`	`Cherebrum`	`I thought it was interesting.`
`02:38.09`	`JT`	`the criticism there isn't that harsh`
`02:38.39`	`Strom_C`	`rudholm: there are plenty people here who will be more than happy to tell you at great length what absolute garbage Asterisk and anything even remotely associated with it is should you choose to ask them :)`
`02:38.58`	`*** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id)`
`02:39.16`	`Cherebrum`	`heh heh`
`02:39.20`	`rudholm`	`no, the criticism isn't very harsh, but it's also not very convincing.`
`02:39.30`	`rudholm`	`I'm running Asterisk without a Digium card and have no timing problems`
`02:39.49`	`rudholm`	`and the criticism says that I am "...likely to experience timing problems."`
`02:39.59`	`Cherebrum`	`rudholm: for personal use or business use?`
`02:40.02`	`JT`	`do you run meetme conferences?`
`02:40.29`	`JT`	`i think the article means without using ztdummy`
`02:40.36`	`rudholm`	`it's personal. but the criticism doesn't make the distinction.`
`02:40.48`	`rudholm`	`and no, I don't run meetme conferences.`
`02:40.50`	`rudholm`	`but that's not my point`
`02:40.59`	`Cherebrum`	`I've had timing issues even with the ztdummy module`
`02:41.00`	`rudholm`	`my point is that the criticism is unconvincing`
`02:41.09`	`Cherebrum`	`RTP gets all wonky and asterisk injects jitter`
`02:41.21`	`rudholm`	`I realize that there are circumstances in which the timing signal from the card is needed`
`02:41.28`	`rudholm`	`and the criticism could have said that`
`02:41.32`	`rudholm`	`and even enumerated those cases`
`02:41.53`	`JT`	`if you aren't happy with it, you could submit an edit`
`02:42.40`	`Cherebrum`	`actually.. the whole criticism part was ripped out. I was looking at the history on the wikipage and I saw that`
`02:42.57`	`Cherebrum`	`also some edit about MArk Spencer not being the primary code maintaner anymore.. What's up with that?`
`02:43.07`	`JT`	`heh, of course it's ripped out`
`02:44.35`	`tzafrir_laptop`	`rudholm, Asterisk requires an external kernel module to operate properly`
`02:44.57`	`rudholm`	`not in all cases.`
`02:45.06`	`Cherebrum`	`doesn't the kernel allready have a timing source?`
`02:45.07`	`rudholm`	`it runs just fine on my server without any.`
`02:45.28`	`JT`	`what the hell does your server do?`
`02:45.38`	`JT`	`i think tzafrir_laptop means for more than a very small volume`
`02:45.40`	`rudholm`	`it's pure VoIP, there are no line cards in it.`
`02:46.03`	`tzafrir_laptop`	`pure voip and no IAX trunks and no meetme`
`02:46.29`	`tzafrir_laptop`	`(unless you use the unofficial app_conference)`
`02:46.30`	`rudholm`	`it uses IAX for call completion (via Teliax)`
`02:46.55`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`02:46.55`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`02:47.57`	`*** join/#asterisk litage (n=nick@203.220.55.70)`
`02:48.02`	`Cherebrum`	`I really like the yate and freeswitch conferences better than the asterisk one anyways. They sound much better and they don't drift like the asterisk ones do`
`02:48.29`	`Strom_C`	`ive never had an asterisk conference drift on me`
`02:49.56`	`Cherebrum`	`There is a noticable diference in audio quality between a ztdummy timed conference and one that is timed by the kernel's exsisting timing source`
`02:50.17`	`Strom_C`	`um`
`02:50.23`	`Strom_C`	`ztdummy uses the kernel's timing source`
`02:50.56`	`Cherebrum`	`then how come it doesn't work as well in a virtual machine?`
`02:50.57`	`*** join/#asterisk codword (n=doc@cpe-65-27-147-15.cinci.res.rr.com)`
`02:51.08`	`rudholm`	`my complaints are with my home Asterisk box' zaptel driver (it often fails to recognize the card on boot)`
`02:51.46`	`codword`	`Hey all... I recently upgraded to fedora core 6 (and.. thusly... got a newer version of asterisk).. while my config files have not changed, my blindtransfer seems to have stopped working... Could anyone point me in a direction to troubleshoot ?`
`02:51.58`	`*** join/#asterisk murdmath (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net)`
`02:52.08`	`Cherebrum`	`codefreeze: check features.conf I think`
`02:52.15`	`Cherebrum`	`er codword:`
`02:52.26`	`Cherebrum`	`damn autocomplete`
`02:52.30`	`codword`	`Yeah? I mean, as I said, my configs have not changed.. it looks fine to me ?`
`02:53.10`	`JT`	`would be nice to say what the version change actually was, codword`
`02:53.31`	`Cherebrum`	`compare the new stock config to yours`
`02:53.37`	`*** join/#asterisk infernix (n=nix@spirit.infernix.net)`
`02:54.13`	`codword`	`JT: 1.2.13 to 1.2.14 I believe`
`02:54.28`	`codword`	`Cherebrum: features.conf is a fairly simple file, its pretty straight forward.. nothing has changed..`
`02:54.34`	`Cherebrum`	`hmm`
`02:54.48`	`codword`	`It's almost like my dmtfmode is wrong for my sip channels.. BUT.. thats not the case because # works fine for things like voice mail functions etc`
`02:55.33`	`codword`	`Note, I am using atrpms.net packages for asterisk... and I'm wondering if perhaps theres a known bug with asterisk-features that they may not have inlucded a patch for..`
`02:56.37`	`codword`	`apparently, you guys are not aware of other people complaining regularly about this.. so ... now I'm worried it's something goofy thats gonna take me forever to find :)`
`02:57.09`	`codword`	`Not sure if me not using zaptel has anything to do with it?`
`02:57.13`	`codword`	`I am 100% VOIP`
`02:57.13`	`JT`	`you should check modules aren't failing at start time`
`02:57.37`	`JT`	`not using zaptel`
`02:57.45`	`JT`	`well of course that could make a differnece`
`02:57.49`	`JT`	`try adding zaptel`
`02:59.07`	`codword`	`zaptel appears to be loading.. but I'm just using the default zapata.conf`
`02:59.10`	`codword`	`I always have....`
`03:00.20`	`clorabit`	`JT: can u help me ? my client can call other extentsion but after i accept call no sound at all is there any wrong wing server configuration ?`
`03:01.04`	`codword`	`codec_zap.c: No Zaptel transcoder support!`
`03:01.07`	`codword`	`is that bad? :)`
`03:01.43`	`JT`	`codword: what's connected to the other extension?`
`03:01.54`	`codword`	`chan_iax2.c: Unable to open IAX timing interface: No such file or directory`
`03:02.00`	`codword`	`those are the only two bad things i notice during the startup`
`03:02.21`	`JT`	`clorabit: i mean`
`03:02.21`	`codword`	`JT: I'm using grandstream BT-102 SIP phones... thats all.`
`03:02.32`	`JT`	`clorabit: what is connected to the other extension?`
`03:02.47`	`codword`	`and everything works great.. and has worked great for a long time... its jsut that my # button seems to have stopped wroking for blind transfer..`
`03:02.54`	`codword`	`oh.. soryr`
`03:03.36`	`demigod2k`	`codword: no offense, but maybe the button broke?`
`03:04.03`	`clorabit`	`JT: i've set 2 extension 1235 and 1234 from 1234 call to 1235 ringing is ok, but after accept call there are no sound delivered from 1234 to 1235`
`03:04.09`	`demigod2k`	`I evaluated those BT's when I was looking for phones. They were somewhere between walmart and "the dollar store" quality`
`03:04.33`	`JT`	`clorabit: umm, use the Echo application before trying to go any further`
`03:04.45`	`codword`	`demigod2k: heh.. multiple phones... also using a linksys PAP-2NA for a few FXS ports, and the # button on the analog phones also quit working...`
`03:05.03`	`codword`	`demigod2k: but I agree, they are cheaply made.`
`03:05.21`	`litage`	`does asterisk rotate its log files automatically?`
`03:05.37`	`JT`	`no`
`03:06.34`	`clorabit`	`JT: echo application also same, i never heard welcome message from echo application, any idea ?`
`03:06.35`	`litage`	`JT: so asterisk admins must either automate it with logrotate, or manually run "logger rotate" every so often?`
`03:07.02`	`JT`	`clorabit: the echo application has no welcome message unless you playback a welcome message first`
`03:07.38`	`JT`	`clorabit: try playing back a sound file you definately have`
`03:07.46`	`*** part/#asterisk Cherebrum (n=jgarland@pdpc/supporter/base/Cherebrum)`
`03:08.00`	`JT`	`Playback(filename) with no .wav or .gsm`
`03:08.26`	`JT`	`since you are using a softphone, it is quite likely your audio settings are incorrectly configured`
`03:09.04`	`JT`	`if you can't playback, or echotest, and they look fine in the console (no errors), you have issues there`
`03:09.15`	`clorabit`	`JT: where i have set this sound file ?`
`03:10.03`	`*** join/#asterisk test34 (n=test34@unaffiliated/test34)`
`03:10.19`	`JT`	`if you don't specify a full path, it will assume the default sounds directory /var/lib/asterisk/sounds/`
`03:10.25`	`*** join/#asterisk Flauto (n=zhao@ppp-68-251-63-122.dsl.chcgil.ameritech.net)`
`03:10.38`	`JT`	`which should have sounds unless you failed to install them`
`03:11.19`	`Flauto`	`i had an error when i tried to make config after compiled and installed asterisk`
`03:11.29`	`Flauto`	`i am using suse 10.2`
`03:12.49`	`clorabit`	`JT: after call echo application i saw that mic meter at idefisk doesn't show any signal ? i've try this client connect to asteriskguru with no problem`
`03:13.17`	`JT`	`does playback have any audio?`
`03:14.01`	`clorabit`	`JT: what u mean with audio ?`
`03:14.09`	`Qwell`	`somebody plz to be helping me with taxes, kthx`
`03:14.26`	`JT`	`clorabit: well when you use the Playback application, can you hear anything??`
`03:14.48`	`Flauto`	`qwell, hehe`
`03:15.05`	`Qwell`	`no, seriously :p`
`03:15.18`	`Flauto`	`get turbotax`
`03:15.22`	`Flauto`	`it is easy`
`03:15.27`	`Flauto`	`i used it last year`
`03:15.31`	`Qwell`	`nah, not mine ;/`
`03:15.41`	`Qwell`	`I went to H&R Block, and it confused them`
`03:15.47`	`Flauto`	`hehe`
`03:15.50`	`Flauto`	`you need a pro`
`03:16.02`	`Flauto`	`hr are for poor people`
`03:16.05`	`Qwell`	`yeah, heh, or somebody who's been in the same situation`
`03:16.10`	`Qwell`	`poor, pfft.. they want $250`
`03:16.25`	`*** join/#asterisk nighty-- (n=nighty-@66-163-28-100.ip.tor.radiant.net)`
`03:16.30`	`Flauto`	`right, they want 250 from poor people`
`03:16.41`	`Flauto`	`rich people don't pay them`
`03:16.48`	`Qwell`	`heh`
`03:17.03`	`Qwell`	`I could still walk away from it, but...meh`
`03:17.19`	`Qwell`	`she's having to consult her bosses, to figure out just htf it's supposed to work`
`03:17.20`	`*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)`
`03:17.36`	`*** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net)`
`03:18.47`	`Flauto`	`hehe`
`03:19.28`	`clorabit`	`JT: is playback app need any parameter to play file ?`
`03:19.43`	`Qwell`	`clorabit: it needs the filename (WITHOUT the extension)`
`03:20.18`	`JT`	`clorabit: i already told you`
`03:20.29`	`JT`	`clorabit: these are pretty basic questions, i suggest you read the book`
`03:20.31`	`JT`	`~thebook`
`03:20.36`	`jbot`	`thebook is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11`
`03:21.44`	`Flauto`	`qwell, i have had a few problems when i reinstalled asterisk on suse 10.2`
`03:23.08`	`Flauto`	`when i did make config after installing asterisk, there was an error`
`03:23.22`	`tzafrir_laptop`	`clorabit, show application playback`
`03:23.26`	`tzafrir_laptop`	`in the CLI`
`03:23.26`	`clorabit`	`JT: ok i've add playback application`
`03:23.32`	`Qwell`	`Flauto: something about not being a supported distro?`
`03:23.52`	`Flauto`	`suse is not supported?`
`03:24.07`	`Flauto`	install: cannot stat `init.asterisk': No such file or directory
`03:24.07`	`Flauto`	`make: *** [config] Error 1`
`03:24.10`	`Flauto`	`this is what i got`
`03:24.13`	`clorabit`	`at asterisk CLI show that Playing hello-world but i don't heard any sound from my client any idea ?`
`03:24.14`	`Qwell`	`oh...umm`
`03:24.23`	`Qwell`	`You aren't using 1.4`
`03:24.32`	`Flauto`	`i am using 1.2`
`03:24.36`	`tzafrir_laptop`	`Flauto, you have debs of 1.2.13 with suse 10.2`
`03:24.44`	`tzafrir_laptop`	`s/debs/rpms/`
`03:24.51`	`Qwell`	`yeah, don't run make config with 1.2, unless you're on RH or something`
`03:25.01`	`Flauto`	`no, i have 1.2.14`
`03:25.05`	`Qwell`	`SuSE is silly...uses silly paths`
`03:25.07`	`Flauto`	`not from rpm`
`03:25.40`	`Flauto`	`so, what should i do`
`03:25.40`	`tzafrir_laptop`	`At least it comes with a proper init script and such`
`03:25.48`	`clorabit`	`JT: at asterisk CLI show that Playing hello-world but i don't heard any sound from my client any idea ?`
`03:25.58`	`Flauto`	`you mean i should get the rpm package?`
`03:26.07`	`JT`	`clorabit: i suggest you switch on your speakers :P although i dunno, maybe there's some other problem`
`03:26.26`	`*** join/#asterisk k-man_ (n=jason@unaffiliated/k-man)`
`03:26.33`	`tzafrir_laptop`	`I don't think suse is that unsupported. I saw some people using asterisk with suse`
`03:26.37`	`JT`	`clorabit: did you Answer the call before doing Playback`
`03:26.48`	`codword`	`very frustrating grrrrrr`
`03:26.53`	`tzafrir_laptop`	`the rpm package is part of the official opensuse distro`
`03:26.54`	`Flauto`	`i mean, everything else seems working fine`
`03:26.57`	`Qwell`	`tzafrir_laptop: it just uses a different init path`
`03:27.09`	`clorabit`	`JT: it is ok when i connect ro asteriskguru servefr`
`03:27.29`	`tzafrir_laptop`	`Qwell, actually, SUSE was the first to implement the LSB standard for init scripts`
`03:27.48`	`Qwell`	`/etc/init.d/?`
`03:27.59`	`k-man_`	`once I have made and installed asterisk, how do i set it up so i can make sip calls on it? is there a guide that will take me from after "make install" to making calls?`
`03:28.07`	`Flauto`	`yes, qwell`
`03:28.08`	`tzafrir_laptop`	`RH places the scripts in /etc/rc.d/init.d`
`03:28.11`	`codword`	`I upgraded to 1.4.0 just for shits and grins.. manually edit each config file to include my necessary configs (which, really, is minor)...`
`03:28.13`	`Flauto`	`it is /etc/init.d`
`03:28.19`	`tzafrir_laptop`	`<PROTECTED>`
`03:28.22`	`codword`	`EVERYTHING works great... except blind transfer! # is still ignored exactly as it was in 1.2.14`
`03:28.25`	`codword`	`w-t-f :(`
`03:28.31`	`Qwell`	`Flauto: You could copy the rc.suse.asterisk from 1.4.. it should work the same`
`03:28.32`	`JT`	`clorabit: version of asterisk, distro and kernel version please`
`03:28.47`	`Flauto`	`hehe`
`03:28.58`	`Flauto`	`so, i need to download the whole thing`
`03:29.08`	`clorabit`	`JT: 1.4, centos 4.3 2.6.9`
`03:29.08`	`Flauto`	`and just to copy it to init.d?`
`03:29.21`	`Qwell`	`nah, you can download the that one file`
`03:29.28`	`JT`	`clorabit: have you tried 1.2?`
`03:29.31`	`Flauto`	`where to?`
`03:29.35`	`tzafrir_laptop`	`Flauto, browse http://svn.digium.com/svn/branches/1.4/`
`03:29.37`	`JT`	`sound like some sort of bug, clorabit`
`03:29.51`	`tzafrir_laptop`	`then copy the link to wget and fetch the file`
`03:29.59`	`Qwell`	`Flauto: get http://svn.digium.com/svn/asterisk/branches/1.4/contrib/init.d/rc.suse.asterisk, and copy it to /etc/init.d/`
`03:30.42`	`clorabit`	`clorabit: no i have not, i can't find any rpm package, is it ok build from source ?`
`03:31.08`	`Flauto`	`okay got it`
`03:31.10`	`Flauto`	`thanks`
`03:31.23`	`JT`	`clorabit: should be find as long as you get rid of the directory of .so files for asterisk first`
`03:31.38`	`Qwell`	`actually`
`03:31.49`	`Qwell`	`Flauto: rename that file to "asterisk", then run /sbin/chkconfig --add asterisk`
`03:32.05`	`Flauto`	`thanks`
`03:32.14`	`Qwell`	`that'll make it start on boot`
`03:32.31`	`Flauto`	`how to change the name`
`03:32.37`	`Bobthehunter`	`anyone got a working 3 way scenraio with openser ?`
`03:32.54`	`Flauto`	`i need to make it excutable as well, i guess`
`03:32.58`	`Qwell`	`Flauto: mv /etc/init.d/rc.suse.asterisk /etc/init.d/asterisk`
`03:32.58`	`Bobthehunter`	`A , B, C where A is ser b,c, asterisk ?`
`03:33.17`	`Bobthehunter`	`if so pm me for paypal ;)`
`03:33.22`	`*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`03:33.49`	`Flauto`	`great`
`03:33.50`	`Flauto`	`thanks`
`03:33.55`	`*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`03:35.00`	`clorabit`	`JT: ok i will try 1.2 then.. thanks`
`03:35.32`	`Qwell`	`so, seriously, who's gonna help me with my taxes? :P`
`03:35.51`	`Qwell`	`please don't make me /j #taxes on efnet, heh`
`03:36.28`	`JT`	`depends how much you want to pay :P`
`03:36.41`	`JT`	`you need to make learning the american taxation system worthwhile`
`03:36.58`	`Qwell`	`heh`
`03:37.04`	`HushPe`	`is there a trick to let me dial an extension number via a zaptel interface (i.e. dial in, press extension number, then it rings) - at the moment it doesn't seem to work i'm using WaitExten`
`03:37.34`	`JT`	`HushPe: DISA allows you full access to a chosen context`
`03:37.42`	`*** join/#asterisk pingwin (n=pingwin@74-138-18-221.dhcp.insightbb.com)`
`03:38.01`	`HushPe`	`ah ok, that might be it :)`
`03:38.26`	`HushPe`	`which would probably let me transfer the calls too right?`
`03:38.46`	`pingwin`	`hey, I've got a simple question. which Asterisk LiveCD is the best?`
`03:38.49`	`HushPe`	`at the moment it just cuts me off`
`03:39.06`	`JT`	`disa gives you a dialtone and drops you into a given context after optionally entering the correct password`
`03:39.16`	`Qwell`	`pingwin: whichever one does what you need`
`03:39.50`	`pingwin`	`Qwell well I'm writing an article on asterisk, and i operate asterisk. however I haven't used any of the liveCD's`
`03:40.09`	`Qwell`	`pingwin: then don't write about something in which you don't know. :)`
`03:40.25`	`Qwell`	`save that behavior for the NY Times and such`
`03:40.40`	`JT`	`and don't promote trixbox/freepbx :P`
`03:40.49`	`pingwin`	`yeah or you can continue to be a prick`
`03:40.53`	`pingwin`	`thanks`
`03:40.57`	`JT`	`err`
`03:41.06`	`JT`	`how is that being a prick`
`03:41.11`	`JT`	`being realistic`
`03:42.02`	`pingwin`	`wwell I can understand it at the aspect of if I am a noob, but I'm not.`
`03:42.10`	`pingwin`	`he doesn't know my skill level.`
`03:42.19`	`Qwell`	`You are a self admitted "noob", when it comes to live CDs`
`03:42.24`	`pingwin`	`he could have given me a straight answer`
`03:42.28`	`JT`	`well i guess he's saying you should test the livecds out`
`03:42.50`	`pingwin`	`no I am not, I said haven't used any of the liveCD's. doesn't mean i'm a noob to liveCD's`
`03:43.03`	`Qwell`	`If I were being a prick, you'd know it`
`03:43.12`	`pingwin`	`or he could have used his even more expert experience than mine, and given me his opinion`
`03:43.27`	`pingwin`	`if he had one`
`03:43.27`	`Qwell`	`If I had an opinion, it'd be me writing the article`
`03:43.38`	`JT`	`noob to asterisk livecds maybe? i dunno`
`03:43.54`	`JT`	`i'm a noob to asterisk livecds`
`03:43.59`	`JT`	`i just don't use them`
`03:44.01`	`Flauto`	`let me try to reboot my computer see if asterisk will start`
`03:44.05`	`Flauto`	`thanks qwell`
`03:44.12`	`Flauto`	`it seems everything went okay`
`03:44.14`	`pingwin`	`neither have I,`
`03:44.23`	`*** join/#asterisk awannabe (n=gti@ip24-251-135-202.ph.ph.cox.net)`
`03:44.34`	`Qwell`	`then you don't have an informed opinion, and you shouldn't be writing about it...`
`03:44.34`	`pingwin`	`but i'm doing a piece on asterisk, and I was going to do a paragraph about "getting your feet wet" with a liveCD`
`03:44.57`	`pingwin`	`and instead of spending a month testing cd's for a paragraph I thought maybe someone in here might know of something worth checking out`
`03:44.58`	`JT`	`pingwin: well, astlinux sounds ok`
`03:45.03`	`codword`	`[Jan 24 22:43:16] WARNING[19063] translate.c: plc_samples 160 format 6`
`03:45.05`	`codword`	`is that bad? :)`
`03:45.10`	`pingwin`	`JT cool, thanks`
`03:45.11`	`awannabe`	`are there any patches to let * have multiple parking lots?`
`03:45.16`	`Qwell`	`a...month? it's a live CD. You can test it in 10 minutes`
`03:45.23`	`JT`	`spend a paragraphs worth of time on it`
`03:45.51`	`JT`	`pingwin: things like trixbox have a gui for noobs, but make it harder to troubleshoot or do advanced tasks with`
`03:45.56`	`*** join/#asterisk AndyCap (n=aoy@pdpc/supporter/sustaining/AndyCap)`
`03:45.59`	`JT`	`and they make bad dialplans`
`03:46.27`	`*** join/#asterisk SethWhit (n=SethW@207-224-14-167.clsp.qwest.net)`
`03:46.34`	`zerowarez`	`hi guys, anyone knows how is the best way to get money with Asterisk ?`
`03:46.39`	`pingwin`	`will note, thanks alot for the advice JT`
`03:46.55`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`03:46.55`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`03:46.57`	`Qwell`	`zerowarez: You could become the next Vonage`
`03:47.16`	`file`	`:D`
`03:47.30`	`greendisease`	`zerowarez: sure. install aterisk onto a machine, then go into your local subway, leave the case open and bang on some cans a get donation from the passersby`
`03:47.42`	`rudholm`	`the best way to get money with Asterisk is to learn it well and do consulting work installing/fixing/configuring it.`
`03:47.45`	`zerowarez`	`aheuhauehaeu`
`03:47.47`	`greendisease`	`it works in new york`
`03:47.52`	`Qwell`	`greendisease: I already said be the next vonage :P no need for a redundant answer`
`03:48.01`	`greendisease`	`Qwell: hahaha`
`03:48.02`	`rudholm`	`heh`
`03:48.17`	`rudholm`	`Qwell: yeah, their IPO was basically that`
`03:48.21`	`Qwell`	`rudholm: totally`
`03:48.23`	`JT`	`but what if you were running the next vonage on that computer that you bang on, at the same time??`
`03:49.00`	`rudholm`	`Qwell: nobody wanted to buy in, so they put giftwrap on the turd and "offered" directed shares to customers as if it was some great privilege`
`03:50.48`	`zerowarez`	`Vonage From Wikipedia, the free encyclopedia Vonage is a commercial voice over IP (VoIP) network and SIP company that provides telephone service via a broadband connection (the company's name is a play on their motto "Voice-Over-Net-AGE").`
`03:51.14`	`zerowarez`	`cool, i'll be a Vonage, thx :D`
`03:51.26`	`Qwell`	`Strom_C: I should've bet`
`03:51.38`	`Strom_C`	`hahahhaa`
`03:51.39`	`Strom_C`	`:D`
`03:52.19`	`zerowarez`	`i'll try to @ least`
`03:52.46`	`zerowarez`	`i''ll study hard first ;)`
`03:53.03`	`rudholm`	`this weekend Strom_C and I saw these mechanical versions of Asterisk. Very interesting how you can actually implement a phone switch in purpose-built hardware.`
`03:53.23`	`rudholm`	`"It's all done with relays"`
`03:53.26`	`Strom_C`	`yeah - something about space-division electromechanical switching`
`03:53.32`	`Strom_C`	`"western electric" or something`
`03:53.42`	`*** part/#asterisk SethWhit (n=SethW@207-224-14-167.clsp.qwest.net)`
`03:53.44`	`*** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net)`
`03:54.01`	`rudholm`	`yeah, and something called "AE" which I think stands for, what was it, "Almost Electric"?`
`03:54.13`	`JT`	`rudholm: what's so unusual about that? :P you can implement a lot of things in purpose built hardware`
`03:54.19`	`Strom_C`	`Almost Equipment :)`
`03:54.24`	`*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)`
`03:54.25`	`JT`	`rudholm: ok, switching trivia time`
`03:54.37`	`Strom_C`	`oh, it's on`
`03:54.45`	`JT`	`rudholm: what was the occupation of the first person to invent an automated telephone switch?`
`03:54.50`	`rudholm`	`oh god`
`03:54.51`	`Strom_C`	`undertaker`
`03:54.53`	`Strom_C`	`duh`
`03:54.57`	`rudholm`	`fucking duh`
`03:54.57`	`Qwell`	`JT: $20 says Strom_C answers every question you have :P`
`03:55.00`	`JT`	`correct`
`03:55.09`	`rudholm`	`complained that the operator was sending calls to his competitor`
`03:55.10`	`Strom_C`	`ask me something DIFFICULT`
`03:55.11`	`Qwell`	`...telco related`
`03:55.12`	`rudholm`	`yeah`
`03:55.24`	`JT`	`i was asking rudholm, but it seems he knows too`
`03:55.28`	`rudholm`	`yes, I know`
`03:55.32`	`rudholm`	`Almon Strowger`
`03:55.36`	`Strom_C`	`oh, rudholm and I will gladly compete`
`03:55.42`	`Strom_C`	`right, rudholm? :)`
`03:55.46`	`rudholm`	`yep`
`03:55.47`	`Nivex`	`this is better than having a trivia bot :)`
`03:56.00`	`JT`	`why was ISDN invented?`
`03:56.08`	`Qwell`	`oh come on`
`03:56.11`	`*** join/#asterisk BSDrAk0 (n=ljd@unaffiliated/luisjose)`
`03:56.12`	`rudholm`	`who invented the phone?`
`03:56.17`	`Qwell`	`rudholm: seriously`
`03:56.19`	`JT`	`bell!`
`03:56.28`	`wunderkin`	`who is picking their nose right now`
`03:56.40`	`rudholm`	`Bell, or Gray???`
`03:56.40`	`JT`	`so why was ISDN invented`
`03:56.49`	`JT`	`Strom_C, rudholm; stop googling!`
`03:57.01`	`Strom_C`	`rudholm would not be googling`
`03:57.03`	`rudholm`	`I don't google`
`03:57.07`	`*** join/#asterisk Flauto (n=zhao@adsl-68-253-252-194.dsl.emhril.ameritech.net)`
`03:57.11`	`JT`	`lies`
`03:57.12`	`Qwell`	`friggen yahoo :p`
`03:57.18`	`rudholm`	`hahaha`
`03:57.26`	`wunderkin`	`ask jeeves`
`03:57.27`	`rudholm`	`Qwell: did a /whois, did we?`
`03:57.33`	`Flauto`	`qwell, another problem, it seems that zaptel was not starting`
`03:57.34`	`Qwell`	`not until just now :)`
`03:57.40`	`Strom_C`	`JT: there was a specific reason beyond "ooh ooh, look, digital telephony!"?`
`03:57.43`	`Qwell`	`rudholm: good guess, I suppose`
`03:57.44`	`Flauto`	`i had to modprobe and ztcfg`
`03:57.57`	`JT`	`do you lose your job if you google at yahoo?`
`03:58.00`	`Qwell`	`rudholm: besides...who uses jeeves?`
`03:58.02`	`Nivex`	`here's one the head of networks at my first employer tried to stump me with:`
`03:58.04`	`Qwell`	`or...msn?`
`03:58.05`	`Nivex`	`In a constellation diagram, all the points lie on a circle centered on the origin. What kind of modulation is being used?`
`03:58.08`	`Nivex`	`if a modem has data points at the following coordinates: (1,1), (1,-1), (-1,1), and (-1,-1).`
`03:58.11`	`Nivex`	`How many bps can a modem with these paramters achieve at 1200 baud?`
`03:58.20`	`Qwell`	`Nivex: 6`
`03:58.39`	`Strom_C`	`Nivex: that's QAM`
`03:58.44`	`Strom_C`	`so therefore....4800`
`03:58.46`	`JT`	`Nivex: 4800bits/s`
`03:58.54`	`JT`	`4 * 1200 omg`
`03:58.56`	`Nivex`	`Strom_C <- the winner`
`03:59.10`	`Strom_C`	`woot`
`03:59.11`	`rudholm`	`1200 baud * four states per baud`
`03:59.21`	`JT`	`i was hesitant in replying because it looked too easy`
`03:59.23`	`Qwell`	`rudholm: welcome to 5 minutes ago`
`03:59.27`	`rudholm`	`haha`
`03:59.32`	`rudholm`	`sorry, I'm actually working`
`03:59.37`	`wunderkin`	`"working"`
`03:59.39`	`rudholm`	`someone has to keep things running here, you know :)`
`03:59.53`	`Qwell`	`rudholm: please, rabid monkeys actually run yahoo`
`04:00.00`	`Qwell`	`that's what slashdot tells me anyhow`
`04:00.06`	`rudholm`	`Qwell: only in senior management`
`04:00.10`	`Nivex`	`just make sure you duck the flying poo`
`04:00.16`	`JT`	`Strom_C: yes, the transmission of NTSC video signals around the US digitally, because phase changes induced in analogue amplifiers cause colour changes in NTSC pictures`
`04:00.28`	`JT`	`(reason for isdn)`
`04:00.37`	`Qwell`	`oh...my...god`
`04:00.37`	`Qwell`	`http://consumerist.com/consumer/stolenidsearch/stolenid-search-see-if--your-idenity-was-stolen-just-type-in-your-ssn-231308.php`
`04:00.47`	`Strom_C`	`JT: what, for picturephone service?`
`04:00.51`	`Qwell`	`you don't even need to read the page - just look at the URL, heh`
`04:00.54`	`JT`	`Strom_C: from tv stations`
`04:00.55`	`JT`	`for`
`04:00.58`	`Strom_C`	`um`
`04:01.01`	`Strom_C`	`that makes little sense`
`04:01.05`	`Strom_C`	`AT&T wasn't in the TV business`
`04:01.06`	`*** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.6.230.revip2.asianet.co.th)`
`04:01.08`	`Flauto`	`qwell, i did modprobe zaptel and wcfxo and ztcfg when i installed zaptel, but when i restarted my computer, i had to re-do modprobe again,`
`04:01.10`	`JT`	`to move video across the US`
`04:01.14`	`JT`	`heh`
`04:01.18`	`BugKhaM`	`any using meetme here?`
`04:01.18`	`Qwell`	`Flauto: you need a zaptel init script also`
`04:01.23`	`Strom_C`	`and anyway, you dont need ISDN for that`
`04:01.29`	`Strom_C`	`just digital transmission facilities`
`04:01.34`	`Flauto`	`from?`
`04:01.35`	`JT`	`tv stations paid the telcos to use it`
`04:01.40`	`rudholm`	`yeah, why would you need ISDN signalling?`
`04:01.45`	`JT`	`yeah and there weren't much of such facilities`
`04:01.58`	`Strom_C`	`that story sounds like 99 and three fifths per cent bullshit to me`
`04:02.11`	`Flauto`	`when i did make config after installing zaptel, i did not get any error though`
`04:02.15`	`BugKhaM`	`I enter meetme with X mode but cannot leave it`
`04:02.16`	`JT`	`which means it's probably true :D`
`04:02.24`	`JT`	`truth is stranger than fiction`
`04:02.29`	`BugKhaM`	`only happened to ZAP calls`
`04:02.30`	`rudholm`	`yeah, I don't buy it`
`04:02.49`	`rudholm`	`I worked for an MPEG member when the MPEG spec was written`
`04:03.17`	`rudholm`	`decent digital video at PRI/T1 speeds didn't exist prior to MPEG-1`
`04:03.35`	`rudholm`	`(the MPEG member in question being Philips, N.V. fwiw)`
`04:03.45`	`rudholm`	`and this was in the early 90s`
`04:03.47`	`rudholm`	`well after ISDN`
`04:04.09`	`JT`	`so you're saying there was no digital video before the 90s?`
`04:04.29`	`rudholm`	`no, I'm saying that there was no good video compression prior to the 90s.`
`04:04.40`	`JT`	`sure`
`04:05.03`	`JT`	`i'm sure the bitrate was high`
`04:05.05`	`*** join/#asterisk joelsolanki (i=joelsola@202.160.161.94)`
`04:05.09`	`rudholm`	`hell, D1 and D2 videotape are from the 80s`
`04:08.27`	`rudholm`	`JT: do you have any references for your ISDN Digital NTSC story?`
`04:08.55`	`JT`	`not on hand`
`04:09.08`	`JT`	`i have papers at home referencing it, from college`
`04:09.15`	`codword`	`WARNING[19294] translate.c: plc_samples 160 format 6`
`04:09.16`	`JT`	`but dunno where they're buried now`
`04:09.20`	`codword`	`Anyone know if that might be some kinda problem?`
`04:09.24`	`codword`	`i cant find anything relevant on google about it`
`04:09.27`	`rudholm`	`do you know where the papers were from? or where they were published?`
`04:10.20`	`JT`	`was years ago, how do you expect me to remember who published them? :) easier just to remember the obscure fact`
`04:10.45`	`rudholm`	`well, I was thinking if it was an AT&T/Bell System doc, you might remember that much`
`04:11.17`	`Strom_C`	`I'm looking in my ISDN book from the early 90s, and it mentions nothing about television transmission`
`04:11.41`	`*** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue)`
`04:11.52`	`JT`	`war on isdn facts ;)`
`04:13.10`	`*** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-246-217-72.washdc.fios.verizon.net)`
`04:13.13`	`Strom_C`	`it does mention video-on-demand as a possible application for ISDN, but in 1990, video-on-demand was one of those cool vaporware things that everyone was talking about`
`04:13.23`	`SomeOne1`	`can someone make a kickass professional looking flash intro for me?`
`04:13.27`	`SomeOne1`	`i'll pay $50 through paypal`
`04:13.42`	`Strom_C`	`interestingly enough, it talks about the future of the ISDN network as what the internet has become instead`
`04:13.45`	`SomeOne1`	`i'm desperate for it, and im good on the payment thing`
`04:13.53`	`SomeOne1`	`but i need it done RIGHT now`
`04:15.25`	`codword`	`hmmmph`
`04:15.26`	`codword`	`well guys....`
`04:15.38`	`codword`	`If anyone else comes on here complaining about transfer not working`
`04:15.45`	`codword`	`Here's the answer for them.`
`04:15.46`	`codword`	`http://bugs.digium.com/view.php?id=8804`
`04:17.51`	`*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)`
`04:18.20`	`rudholm`	`yeah`
`04:18.35`	`rudholm`	`we'll make a note of it`
`04:18.40`	`Strom_C`	`"B-channels operate at 64kbps, but user equipment does not always generate data at that rate. For example, a personal computer in an office may not be able to operate at speeds greater than 9600bps."`
`04:18.48`	`rudholm`	`heh`
`04:18.57`	`JT`	`hah`
`04:19.01`	`rudholm`	`good thing we have all that headroom`
`04:19.05`	`rudholm`	`we'll never outgrow that`
`04:19.27`	`Strom_C`	`who would ever possibly want to transmit at 64kbps?`
`04:19.57`	`JT`	`9600 would fit in the d channel`
`04:20.07`	`rudholm`	`GSM was originally intended to be full wireless ISDN (at BRI speeds)`
`04:20.30`	`Strom_C`	`now that would be the sex`
`04:20.38`	`rudholm`	`yeah`
`04:20.39`	`JT`	`that idea clearly died in the arse ;)`
`04:20.40`	`*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`04:20.50`	`rudholm`	`yeah, they realized they couldn't get that much bandwidth`
`04:20.54`	`*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`04:21.08`	`rudholm`	`but the architecture is the same, they just reduced the B channels down to 4800bps`
`04:21.11`	`rudholm`	`dunno about the D channel`
`04:21.19`	`JT`	`what is with mtaht4's join/parts`
`04:21.31`	`JT`	`there's quite a lot of different types of control timeslots in gsm`
`04:25.37`	`BugKhaM`	`DTMF doesn't work with ZAP in meetme`
`04:25.37`	`BugKhaM`	`anyone knows how to fix this?`
`04:25.51`	`codword`	`http://bugs.digium.com/view.php?id=8804`
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`04:37.03`	`*** join/#asterisk Flauto (n=zhao@ppp-68-251-57-249.dsl.chcgil.ameritech.net)`
`04:37.39`	`Flauto`	`qwell, would you send me the link for zaptel init script`
`04:37.45`	`Qwell`	`it's in zaptel`
`04:38.01`	`Flauto`	`1.2?`
`04:38.06`	`Qwell`	`should be fine`
`04:38.13`	`Flauto`	`okay`
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`04:45.25`	`drray`	`Hi, I'm getting red alarms on my adit 600, it's set via dip switches and has worked fine for a year and change. I removed it from the setup (a tormenta III card) and tested it on another box with a wildcard t100p on it. The fxs lights are flashing orange then greeen, orange then green.`
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`04:49.00`	`tzafrir_laptop`	`red alarm sounds like no wire or something...`
`04:49.52`	`*** part/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net)`
`04:50.41`	`*** join/#asterisk Flauto (n=zhao@ppp-68-251-47-75.dsl.chcgil.ameritech.net)`
`04:50.56`	`Flauto`	`qwell, still working working`
`04:51.10`	`Flauto`	`as soon as i did modprobe, asterisk started automatically`
`04:51.21`	`Flauto`	`so, it is the modprobe step which is missing`
`04:51.45`	`bkw_`	`Got Speech?`
`04:51.48`	`bkw_`	`oh Qwell`
`04:51.54`	`bkw_`	`oh where art thou?`
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`04:56.49`	`xpot`	`is there a way to get a partial cdr? For example, a caller calls in and hangs up after a few seconds of connect (or any time during the call before the connect to user)`
`04:59.44`	`bkw_`	`you should get a CDR for that`
`04:59.48`	`bkw_`	`if not then something is bbbbbbbroken`
`05:01.49`	`CunningPike`	`xpot: That is the difference between 'duration' and 'billableduration' - for a call such as you describe, a CDR is created with a billable duration of 0`
`05:02.14`	`CunningPike`	`Flauto: Which distro?`
`05:02.24`	`Flauto`	`suse 10.2`
`05:02.52`	`tzafrir_laptop`	`Flauto, which hardware?`
`05:02.57`	`CunningPike`	`Flauto: We're on RHEL, but I know I had to modify 1.2 init scripts to contain the paths to modprobe and other bins`
`05:03.00`	`Flauto`	`wcfxo`
`05:03.21`	`CunningPike`	`Flauto: As in '/sbin/modprobe foo' etc`
`05:03.30`	`tzafrir_laptop`	`doesn't wcfxo getmodprobed automatically?`
`05:03.38`	`Flauto`	`no`
`05:03.47`	`Flauto`	`it does not for some reason`
`05:04.15`	`Flauto`	`what you mean? qunninglike`
`05:04.27`	`tzafrir_laptop`	`remove all the silly "install" lines from /etc/modprobe.d/zaptel and use a propr /etc/init.d/zaptel script`
`05:04.53`	`Flauto`	`let me see`
`05:06.17`	`Flauto`	`there is a lot of stuff in there`
`05:07.56`	`Flauto`	`you mean remove everything there in /etc/modprobe.d/zaptel?`
`05:08.36`	`Flauto`	`there is only one i need, which is wcfxo`
`05:08.47`	`tzafrir_laptop`	`rem-out, with a #`
`05:09.01`	`Flauto`	`all of them?`
`05:09.15`	`tzafrir_laptop`	`just the scfxo is the one that affects you`
`05:09.20`	`tzafrir_laptop`	`wcfxo`
`05:09.34`	`Flauto`	`leave along only that one`
`05:10.08`	`Flauto`	`and use # for the others?`
`05:11.15`	`Flauto`	`just did it`
`05:11.45`	`Flauto`	`what do i need to do to the script at /etc/init.d/zaptel then?`
`05:13.23`	`Flauto`	`or, i am done?`
`05:15.45`	`Flauto`	`tzafrir, are you there`
`05:16.46`	`*** join/#asterisk PuTzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)`
`05:16.50`	`tzafrir_laptop`	`don't you have one?`
`05:18.56`	`Flauto`	`i have one there`
`05:18.57`	`greendisease`	`hey so any idea when beta4 of *now is due?`
`05:19.39`	`Flauto`	`okay, i think that is gonna help`
`05:19.40`	`Flauto`	`thanks`
`05:20.00`	`k-man_`	`do many people use FWD?`
`05:20.40`	`Flauto`	`let me try it`
`05:20.47`	`Flauto`	`thanks, tzafrir`
`05:23.55`	`PuTzz`	`30,783 FWD users online at this time looks like they do`
`05:24.14`	`*** join/#asterisk HushPe (n=HushPe@202.36.75.196)`
`05:25.34`	`HushPe`	`JT: just clarifying about DISA, do i set my Zap group to call a context which answers using DISA? then when a user dials an extension, they can be forwarded to it? i.e. ring ring, answer, disa > context > extensions > dial extension (SIP mostly)`
`05:26.00`	`k-man_`	`putzz, how does FWD make money? any idea?`
`05:26.17`	`HushPe`	`so that means that WaitExten will work from inside the PBX, but not when dialing in using a Zap card?`
`05:28.21`	`PuTzz`	`FWD why would they need to make money? they dont have outgoing except for tollfree numbers`
`05:28.39`	`PuTzz`	`FWD = pulver communications wich makes money`
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`05:38.18`	`jlimb`	`Is there anyway to determine if a phone is off hook?`
`05:38.39`	`CunningPike`	`jlimb: Yup - look at it`
`05:38.47`	`CunningPike`	`jlewis: Ba-bum-tish`
`05:38.52`	`jlimb`	`smart :)`
`05:38.56`	`CunningPike`	`:D`
`05:39.02`	`jlimb`	`lol`
`05:39.22`	`*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)`
`05:39.24`	`CunningPike`	`jlimb: What are you trying to do?`
`05:39.25`	`blitzrage`	`evening all!!!`
`05:39.27`	`blitzrage`	`hey, if I use one-touch recording (wW flag in Dial()), is there a way to get a tone indicating recording has started?`
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`05:41.15`	`blitzrage`	`Juggie: any ideas?`
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`06:01.22`	`murdmath`	`Does asterisk have a "state" for when the handset is off hook, nothing has been dialed?`
`06:03.01`	`niZon`	`nothing thats really executed in the dialplan...`
`06:03.17`	`niZon`	`chan_sccp/skinny knows when a phone is off hook, but it doesn't do much`
`06:03.55`	`sevard`	`if you have something like a tdm2400p run asterisk in a very verbose mode, or zaptel debug, i can't remember which, but you can see when somebody picks up the phone.`
`06:04.00`	`niZon`	`some devices have a hotline feature, you could have it auto dial into * then have * handle the dialing`
`06:05.07`	`murdmath`	`We are trying to solve a problem with paging.`
`06:05.33`	`murdmath`	`If someone has the handset off the hook, and a page comes in, it actually rings the phone.`
`06:06.59`	`murdmath`	`We are working with snom phones.`
`06:07.00`	`*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)`
`06:07.41`	`*** join/#asterisk yxa (n=lonari@58.185.90.101)`
`06:08.15`	`yxa`	`can I group misdn channels? so that it will search for a free line out of 4 ports?`
`06:08.55`	`JT`	`only on TE ports i believe`
`06:12.25`	`[TK]D-Fender`	`murdmath : That falls under the category of TFB`
`06:17.06`	`*** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net)`
`06:20.17`	`yxa`	`JT how do i do it?`
`06:22.04`	`murdmath`	`[TK]D-Fender: TFB? (pardon my ignorance)`
`06:23.47`	`JT`	`yxa: just like in zaptel`
`06:24.34`	`[TK]D-Fender`	`~tfb`
`06:24.35`	`jbot`	`i guess tfb is Too #&^$ing bad....`
`06:25.47`	`[TK]D-Fender`	`murdmath : SIP phones have their own "brains" and tell things to the server when THEY feel like it, when it has something to say in the scope of what SIP is meant to be able to do. "Off-hook" in NOT part of that.`
`06:26.28`	`[TK]D-Fender`	`murdmath : You will usually get that with analog Zaptel channels, and MGCP, and Skinny/SCCP (I believe)`
`06:29.49`	`[TK]D-Fender`	`Ok, checkout time... later all...`
`06:30.00`	`*** join/#asterisk joelsolanki (i=joelsola@202.160.161.94)`
`06:30.25`	`yxa`	`port1 of my misdn is assign 8 extensions from my telco. now that i want to add another 8 ext to port2, they tell me they cant do a huntgroup. are they on crack?`
`06:30.46`	`BugKhaM`	`anyone using meetme with ZAP clients at all?`
`06:31.47`	`yxa`	`their exact words: "The prob with ISDN-2 is that the MSN nos cannot be shared across another ISDN-2; One way to work around this is to split the nos (8 MSN on 1 ISDN-2, and another 8 MSN on another ISDN-2); Other option would be to upgrade to an ISDN-10, with a whole new set of DDI nos.."`
`06:32.07`	`yxa`	`is that true?`
`06:34.06`	`*** join/#asterisk oej (n=olle@apollo.webway.se)`
`06:34.27`	`JT`	`no`
`06:34.30`	`JT`	`that is false`
`06:34.33`	`JT`	`your telco is an idiot`
`06:35.08`	`yxa`	`JT not surprised you said that`
`06:35.28`	`yxa`	`JT how should i "teach" them to do?`
`06:37.10`	`JT`	`maybe there's an 8 extension limit`
`06:37.23`	`JT`	`but technically, it think you can get a lot more lines`
`06:37.31`	`JT`	`there's no isdn technical limitation`
`06:37.39`	`JT`	`as long as they send full digits in the msn`
`06:37.45`	`JT`	`your end can pick it up`
`06:37.58`	`yxa`	`they do send full digits in the msn as of now.`
`06:38.03`	`JT`	`unless they're hardware or software is crap`
`06:38.05`	`*** join/#asterisk MrY (n=silencer@adsl-71-146-131-183.dsl.pltn13.sbcglobal.net)`
`06:38.12`	`JT`	`i'm sure it can be done`
`06:38.22`	`MrY`	`hi all`
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`06:39.13`	`MrY`	`i have a p4 2.8ghz, how many extensions can my box handle simultaneously? ie on the phone...`
`06:40.10`	`J4k3`	`MrY: depends on a lot of variables. I'd estimate anywhere from 20 to 2,000.`
`06:40.15`	`JT`	`depends how they're connected, but probably a few`
`06:40.21`	`JT`	`and if you're transcoding`
`06:40.24`	`nilkanthp`	`MrY, Well I have p4 with 2GB RAM and I run 80 extensions`
`06:40.44`	`sevard`	`RAM is an issue. also, what matters more is not sym calls but transcoding`
`06:40.52`	`sevard`	`damn wireless lag, got beaten`
`06:41.07`	`MrY`	`nilkanthp: can you box handle all 80 extensions actively connected (talking)`
`06:41.30`	`nilkanthp`	`yes, very well`
`06:42.29`	`yxa`	`JT is there any documentation on the net i can show them?`
`06:42.44`	`MrY`	`i have a p4 2.8, but experience loads in 2.0 to 4.0 with just 5 extensions on the phone... the daemon mpg123 is taking up lot of resources.. not sure why.`
`06:44.01`	`yxa`	`JT but it doesnt make sense. If both ports listen to a same extension, would it be confused as to who will pick it up?`
`06:44.27`	`JT`	`what?`
`06:44.33`	`JT`	`what do you mean`
`06:45.44`	`yxa`	`JT now that I have only 1 set of 8 ext, when i add another 8, both ports will be listening to the 16 ext?`
`06:46.06`	`JT`	`you said they'd only let you have 8 different ones on each port`
`06:46.36`	`yxa`	`yeah`
`06:46.55`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
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`06:47.10`	`JT`	`so why would both ports be listening for the same extension?`
`06:47.18`	`yxa`	`but i'm not applying them at the same time. we got 8 last year. and this year we are going to add another 8`
`06:48.11`	`yxa`	`so when the first port are in used, incoming calls to the first 8 will get a busy tone. we want to utilize the 2nd port too`
`06:48.26`	`JT`	`yes that's a telco issue`
`06:48.31`	`JT`	`not asterisk`
`06:49.13`	`yxa`	`JT i know. so what are they _supposed_ to do?`
`06:49.19`	`yxa`	`a hunt group?`
`06:49.24`	`JT`	`yeah`
`06:49.27`	`JT`	`a linehunt group`
`06:49.54`	`yxa`	`for all 16 numbers?`
`06:49.59`	`JT`	`yes`
`06:50.04`	`yxa`	`or just 1 from each set`
`06:50.38`	`JT`	`err, they should be able to put them all in a linehunt`
`06:50.53`	`yxa`	`you are sure they can do that for bri numbers? i'm gonna confront them on that`
`06:51.37`	`JT`	`here in .au, i have a system running 3 BRIs sharing a few numbers in a single linehunt group`
`06:51.50`	`JT`	`of course, if it's a residential blan, they'd probably refuse to do it`
`06:52.23`	`*** join/#asterisk susinths (n=susinths@trollhoin.ifi.uio.no)`
`06:54.30`	`yxa`	`dammit`
`06:55.20`	`JT`	`you're on a residential plan?`
`06:55.35`	`*** join/#asterisk topping (n=topping@204.152.96.50)`
`06:55.56`	`yxa`	`no, but its a pretty cheap plan with non-running numbers`
`06:56.09`	`JT`	`i see`
`06:56.10`	`yxa`	`can you believe it? 8 NON RUNNING NUMBERS for a business`
`06:56.20`	`yxa`	`fuck`
`06:56.20`	`JT`	`hmm`
`06:57.36`	`yxa`	`JT when you got your 3 bris did you get them all at the same time?`
`06:57.56`	`JT`	`yes`
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`06:58.14`	`yxa`	`maybe that's why`
`06:58.46`	`JT`	`i dunno`
`06:59.49`	`yxa`	`anyone else here has a 2nd opinion?`
`07:00.16`	`JT`	`what country are you in?`
`07:00.34`	`yxa`	`singapore`
`07:01.11`	`JT`	`how many bris do you have?`
`07:01.26`	`yxa`	`as of now 1`
`07:01.55`	`JT`	`i thought you had 2`
`07:02.05`	`JT`	`why do you need so many numbers for so few lines?`
`07:02.29`	`yxa`	`i have 1 and going to get another. its not my call.`
`07:02.47`	`JT`	`do you need so many numbers`
`07:04.24`	`yxa`	`as it said its not my call. i'm just the engineer\`
`07:04.52`	`JT`	`somtimes the engineer has to report back to management that it is not possible`
`07:06.21`	`JT`	`and suggest new solutions, suggesting new solutions requires understanding the business need`
`07:07.53`	`yxa`	`lets just assume that we need 16 DIDs for 16 staff`
`07:08.16`	`JT`	`it could work with 8 on each bri, just not as well`
`07:08.32`	`JT`	`people will get busy tones during high call volume on the bri (2 calls)`
`07:09.13`	`yxa`	`yeah, precisely why i wanna add another`
`07:09.32`	`*** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net)`
`07:09.44`	`JT`	`something you'll have to negotiate with your telco`
`07:11.21`	`yxa`	`yeah. and thanks for the comments JT`
`07:12.17`	`JT`	`i find bristuff a little better than misdn (or a lot better, depending on your circumstances), but if misdn works, no real reason to switch from it`
`07:15.58`	`yxa`	`misdn works pretty ok for me`
`07:16.00`	`yxa`	`so far`
`07:16.28`	`JT`	`if you do basic stuff with it, and use only single port cards, it can work ok`
`07:25.02`	`*** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell)`
`07:25.02`	`*** mode/#asterisk [+o Qwell[]] by ChanServ`
`07:25.44`	`data23`	`yawns`
`07:26.56`	`*** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53)`
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`07:31.23`	`yxa`	`JT you still there?`
`07:31.27`	`JT`	`yes`
`07:32.42`	`yxa`	`JT what if I do not add any more extensions. just the bri. can it be done?`
`07:32.56`	`JT`	`can what be done?`
`07:33.16`	`JT`	`putting all MSNs on both BRIs? that's up to the telco`
`07:33.24`	`yxa`	`yeah`
`07:33.49`	`yxa`	`putting x MSNs on y BRIs`
`07:33.52`	`JT`	`telco must implement that for incoming calls`
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`07:41.58`	`robin_sz`	`yxa, wait until you try DTMF tone detection with misdn :)`
`07:43.04`	`robin_sz`	`yxa, works for some phones, not others ... especially callers with older analogue phones, misdn doesnt seem to detect the dtmf digits reliably`
`07:43.49`	`gfraysse`	`<PROTECTED>`
`07:45.54`	`*** join/#asterisk burus (n=burus@host-86-106-208-182.moldtelecom.md)`
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`07:46.55`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`07:48.56`	`*** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net)`
`07:49.16`	`*** part/#asterisk burus (n=burus@host-86-106-208-182.moldtelecom.md)`
`07:50.26`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`07:51.09`	`Defraz`	`Does anyone know if there is a codec for h264 for askerisk 1.2.`
`07:51.13`	`Defraz`	`I understand it is in 1.4`
`07:51.30`	`Defraz`	`I am trying to get two Grandstream gvx 3000 phones working`
`07:51.48`	`Defraz`	`but it only supports g264 and what I have read nobody has anything for it on asterisk.`
`07:51.52`	`Defraz`	`Just was curious`
`07:52.10`	`JT`	`oh dear, grandstream make video phones now?`
`07:52.43`	`hads`	`They have for some time now`
`07:52.50`	`JT`	`hmm`
`07:53.06`	`hads`	`It's a bit ugly but aparently it works.`
`07:56.07`	`zoa`	`Defraz: no there is not but passthrough should work`
`07:56.14`	`zoa`	`let me check if i have a tutorial for it`
`07:57.13`	`zoa`	`http://www.asteriskguru.com/tutorials/gxv_3000.html`
`08:03.46`	`*** join/#asterisk UlbabraB (n=salama@ip-204-57.sn2.eutelia.it)`
`08:04.35`	`*** join/#asterisk reber (i=reber@gateway/tor/x-eacd3077f2649ae4)`
`08:06.14`	`*** join/#asterisk greendisease (n=jack@fedora/greendisease)`
`08:10.10`	`*** join/#asterisk anthonyl (n=anthonyl@adsl-230-23-246.hsv.bellsouth.net)`
`08:10.21`	`*** join/#asterisk HushPe (n=HushPe@202.36.75.196)`
`08:11.17`	`HushPe`	`in my dial plan, i have an incoming and then a officehours contexts. do i need exten => s,1,Answer() in the office, or just exten => s,1,Dial(${MYPHONES}) ?`
`08:11.53`	`HushPe`	`i know i need to answer in the incoming :)`
`08:12.47`	`*** join/#asterisk UlbabraB (n=salama@ip-204-57.sn2.eutelia.it)`
`08:14.04`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`08:15.20`	`osiris`	`may i take a general poll ?`
`08:15.51`	`HushPe`	`what are you polling?`
`08:16.00`	`osiris`	`what are people paying for ip trunks, and from whoooooo`
`08:16.10`	`osiris`	`sorry about the o's`
`08:16.15`	`osiris`	`this keyboard`
`08:17.00`	`Strom_C`	`whooooooooo is perfectly acceptable if I get to go on and on about level threeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee`
`08:17.17`	`osiris`	`yes, level 3 is very kewl`
`08:17.32`	`Strom_C`	`eeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee`
`08:17.36`	`Strom_C`	`ok, i'm done :)`
`08:17.38`	`J4k3`	`e!`
`08:17.59`	`HushPe`	`here in nz there are a few providers, at work we use the local wireless isp, it's free connection and calls within their networks are free, outgoing calls are 5c a min (NZD), and mobiles are quite a bit more like 45c (NZD) a min`
`08:18.11`	`zoa`	`noooooooooooooooooooooooooooooooooooo prooooooooooooooooooooooooooooooooooooblem fooooooooooooooooooooooooooooooooooor the ooooooooooooooooooooooooooooooooooooo's`
`08:19.23`	`osiris`	`i guess i meant ip trunk registrations with a voip provider`
`08:20.14`	`osiris`	`concurent calls, and # of registrations. did's.....`
`08:20.51`	`HushPe`	`dids are about $25/mo (all nzd), i think we only have one registration, but i'm sure it's not too much of a hassle to get more`
`08:21.24`	`HushPe`	`we use ours mostly for outgoing calls`
`08:22.14`	`osiris`	`from what country`
`08:22.30`	`*** join/#asterisk hellop1 (n=hellop@udp115314uds.hawaiiantel.net)`
`08:31.56`	`*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)`
`08:38.38`	`*** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net)`
`08:42.01`	`*** join/#asterisk psk (n=psk@golia.caltanet.it)`
`08:48.36`	`hads`	`HushPe: NZ eh?`
`08:48.46`	`HushPe`	`hads: indeed`
`08:48.58`	`hads`	`Where abouts are you?`
`08:49.01`	`HushPe`	`phone services are ratshit here`
`08:49.02`	`HushPe`	`tauranga`
`08:49.04`	`HushPe`	`you?`
`08:49.07`	`hads`	`Timaru`
`08:50.09`	`*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)`
`08:50.25`	`HushPe`	`you're not nicegear are you?`
`08:50.37`	`hads`	`Yeah`
`08:50.45`	`HushPe`	`i'm arron from KAMAR`
`08:50.55`	`hads`	`Oh. Hi :)`
`08:51.02`	`HushPe`	`organizing our voice server so the boss buys more stuff (it was my idea)`
`08:51.18`	`hads`	`Cool`
`08:51.24`	`hads`	`How's it all going?`
`08:51.29`	`HushPe`	`i do helpdesk there, sick of telecom's voicemail man, press 1 blah, i like it emailed to me :)`
`08:52.05`	`HushPe`	`getting there, initial problems with the IRQs, but i got some help in here and put noapic in the kernel options, which solved my sharing and crackly problem, now crystal clear`
`08:52.17`	`hads`	`Excellent`
`08:52.30`	`HushPe`	`problem now is when i ring to the land line, asterisk doesn't pick up the line (like open it up)`
`08:52.41`	`HushPe`	`so start playing the welcome message, but it keeps on ringing`
`08:52.54`	`HushPe`	`have a GSM box which it works perfectly with (erricson one)`
`08:52.58`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`08:53.29`	`HushPe`	`so i'm just setting up our dial plan now (at home) to test tomorrow, and hopefully i can work out the pick problem with the telecom line`
`08:54.14`	`hads`	`So what's the problem? You call into the analog line from an external PSTN line.`
`08:54.29`	`HushPe`	`yeah, it rings but doesn't pick up`
`08:54.39`	`HushPe`	`call out works a charm :)`
`08:54.48`	`hads`	`Interesting`
`08:55.03`	`Chris-NB`	`hi`
`08:55.10`	`hads`	`Do you get output on the console to say that the line is ringing?`
`08:55.15`	`HushPe`	`yep`
`08:55.24`	`HushPe`	`then i get warnings`
`08:55.31`	`Chris-NB`	`anyone testet Q.SIG signaling bewteen Asterisk and an Alcatel PBX?`
`08:55.33`	`HushPe`	`i don't have them with me at the moment`
`08:55.34`	`hads`	`What are the warnings?`
`08:55.40`	`hads`	`Oh, bummer`
`08:55.52`	`HushPe`	`something about state 6`
`08:57.17`	`HushPe`	`hang on i might have it in a log`
`08:57.49`	`*** join/#asterisk qdk (n=qdk@213.150.62.32)`
`08:58.14`	`HushPe`	`chan_zap.c: Ring/Off-hook in strange state 6 on channel 1`
`08:58.23`	`HushPe`	`then it ring again`
`08:58.26`	`HushPe`	`and does that again`
`08:58.40`	`HushPe`	`then when i hang up the PSTN i get`
`08:58.48`	`HushPe`	`Jan 25 10:12:52 NOTICE[9111] chan_zap.c: Got event 17 (Polarity Reversal)...`
`08:59.10`	`HushPe`	`which if i answer i get a dial tone (telecom) and can happily dial out if it really felt the urge`
`08:59.40`	`*** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk)`
`08:59.41`	`HushPe`	`so i'm guessing we have call forward reversal on, as our current pbx (hybrex) works fine with it`
`08:59.55`	`hads`	`Yeah, looks like it`
`08:59.57`	`HushPe`	`(afk shower, i'll check back when i get out)`
`09:00.06`	`HushPe`	`i have fksks i think`
`09:00.19`	`HushPe`	`which works good for the gsm channel`
`09:00.31`	`hads`	`Yeah that's correct`
`09:00.36`	`HushPe`	`yep`
`09:00.47`	`HushPe`	`so i'm not too sure why it's not actually picking up correctly`
`09:01.18`	`HushPe`	`i even get caller id from the gsm box ;)`
`09:02.11`	`*** part/#asterisk jlimb (n=user@networkv.dsl.xmission.com)`
`09:03.41`	`hads`	`What's your incoming context look like?`
`09:03.44`	`*** join/#asterisk reber (i=reber@gateway/tor/x-a981856b7f51e8f3)`
`09:07.08`	`hads`	`and your modprobe options`
`09:12.09`	`ThoMe`	`Good Morning.`
`09:15.55`	`*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)`
`09:20.34`	`*** join/#asterisk HushPe (n=HushPe@202.36.75.196)`
`09:21.01`	`HushPe`	`hads: i'm about to head off to bed (5:30 start), will you be around tomorrow`
`09:21.45`	`*** join/#asterisk Brumle (n=brumle@brumle.com)`
`09:22.03`	`HushPe`	`modprobe options are the default (so everything i think), incoming context is about to be changing, but it tired an answer, welcome message, then dial extension you want, or answer and dial a group of sip/zap phones`
`09:22.53`	`hads`	`Check on the astug.org.nz site for the nz modprobe options`
`09:23.01`	`HushPe`	`oh, i added those too :)`
`09:23.07`	`HushPe`	`down the bottom`
`09:23.24`	`hads`	`Down the bottom of what?`
`09:23.36`	`HushPe`	`of the modprobe.d/zaptel file`
`09:23.49`	`HushPe`	`i see it setting nz mode in the dmesg`
`09:24.05`	`ThoMe`	`Na, spricht hier auch wer deutsch?`
`09:24.29`	`hads`	`OK that's good. I'd suggest a post to the astug.org.nz mailing list with these details.`
`09:25.09`	`HushPe`	`ok, i hadn't got a change to do that yet, i found in here pretty helpful, but i'm guessing there are some very nz specific settings`
`09:25.17`	`HushPe`	`with telecom and all ;)`
`09:25.48`	`hads`	`Yeah, the settings aren't too specific. I just suggest that as it's the place I keep an eye on the most.`
`09:26.00`	`HushPe`	`ah yep`
`09:26.25`	`HushPe`	`i must say the snom phone is really great! web interface, but in terms of voip phone hardware i don't think you could get too much better`
`09:27.15`	`hads`	`Yeah, I quite like the snoms`
`09:27.32`	`hads`	`So you see something like this in your dmesg? Module 0: Installed -- AUTO FXO (NEWZEALAND mode)`
`09:27.35`	`HushPe`	`i knew they were linux based that's why i had the boss get those`
`09:27.44`	`HushPe`	`yep that's the one`
`09:27.48`	`hads`	`Cool`
`09:30.33`	`hads`	`HushPe: I get the strage state 6 message sometimes and it doesn't seem to do a lot.`
`09:30.48`	`hads`	`It is only a warning, not an error.`
`09:30.51`	`HushPe`	`yeah`
`09:31.00`	`HushPe`	`only odd thing for me it it doesn't pick up the line`
`09:31.18`	`HushPe`	`initially i thought it was the irq and fixing that would resolve it, but no avai`
`09:31.41`	`HushPe`	`avail*, but could be my context too, so hopefully with the new one i'm doing it may resolve it`
`09:32.06`	`hads`	`Yeah, a problem with the context is what I'm thinking at the moment.`
`09:32.11`	`hads`	`Until proved otherwise.`
`09:32.53`	`HushPe`	`yep, well off to sleep i go... i'll pastie/pastebin my extensions.conf tomorrow hopefully some good feedback will come from here`
`09:33.06`	`hads`	`Later mate`
`09:33.08`	`HushPe`	`apparently my current one is insecure, which it no surprise it was my 'get it going' one`
`09:33.15`	`Chris-NB`	`can someone plz look at that: http://pastebin.ca/327845`
`09:33.52`	`Chris-NB`	`If I call 436621234 why the 2nd block is choosen? not the 3rd one?`
`09:34.29`	`HushPe`	`Chris-NB: you might need to check the ordering think on the asterisk wiki`
`09:34.35`	`HushPe`	`i recall reading something on there about regex ording`
`09:35.00`	`Chris-NB`	`HushPe, I've allready changed the order, same result`
`09:35.12`	`*** join/#asterisk Avochelm (n=damien@gw-morphett.koalatelecom.com.au)`
`09:35.15`	`*** join/#asterisk frogzoo (n=frogzoo@202.155.165.25)`
`09:35.23`	`*** join/#asterisk frogzoo (n=frogzoo@202.155.165.25)`
`09:35.32`	`hads`	`The order doesn't matter, you need to specify them in different contexts and include them`
`09:35.41`	`Chris-NB`	`HushPe, I thought, the tighter match is user. so 43662 should match more than 43. right?`
`09:35.59`	`Chris-NB`	`hads, different contexts? why?`
`09:36.23`	`HushPe`	`there you go :)`
`09:36.30`	`HushPe`	`something about the contexts order which does it`
`09:36.37`	`HushPe`	`includes are done last or something rihgt?`
`09:36.47`	`hads`	`Correct`
`09:36.51`	`Chris-NB`	`ok`
`09:36.55`	`HushPe`	`night`
`09:37.00`	`*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)`
`09:37.20`	`Chris-NB`	`but the regex which matches most is used. Isn't that true?`
`09:38.37`	`hads`	`Chris-NB: It's getting late here. Check the wiki, there is a page with explains it there.`
`09:38.46`	`hads`	`s/with/which/`
`09:40.10`	`Chris-NB`	`k, just reading it up in the asterisk book`
`09:45.52`	`Chris-NB`	`the asterisk book says, the more specific regex is used`
`09:46.50`	`*** join/#asterisk Jenna (i=CherryRe@gateway/tor/x-9d858b76489992de)`
`09:46.51`	`Chris-NB`	`so for 436621234 this one should be used: _43662. and not that: _43. <-- right?`
`09:48.42`	`Jenna`	`hi guys, Anyone with experience of using sangoma cards especailly http://www.sangoma.com/datasheets/p_a200-specs the analog telephony card with asterisk ?`
`09:49.10`	`Chris-NB`	`hads, found it out. I've to put these two extensions in: _43X. and _43662X.`
`09:49.25`	`Chris-NB`	`than the more specific (_43662X.) is used.`
`10:06.35`	`JT`	`why are you doing funny stuff like that? :)`
`10:06.49`	`Chris-NB`	`me?`
`10:06.55`	`JT`	`yeah`
`10:07.07`	`JT`	`do you need them to be handled differently?`
`10:07.11`	`Chris-NB`	`cause if to decide if its a local or national call`
`10:07.29`	`JT`	`what is 43?`
`10:07.38`	`Chris-NB`	`international prefix for AT`
`10:07.42`	`Chris-NB`	`0043`
`10:07.49`	`Makenshi`	`hmm,, denmark iirc`
`10:07.52`	`Chris-NB`	`nop`
`10:07.56`	`Chris-NB`	`Austria == AT`
`10:07.56`	`JT`	`at is austria`
`10:08.06`	`Makenshi`	`oopsie`
`10:08.10`	`JT`	`at sounds nothing like denmark lol`
`10:08.11`	`Chris-NB`	`: ) np`
`10:08.15`	`Chris-NB`	`*hrhr`
`10:08.17`	`Makenshi`	`45 is denmark`
`10:08.23`	`Makenshi`	`i thought you were talking modem speak`
`10:08.24`	`Chris-NB`	`dunno`
`10:08.33`	`JT`	`so what is 662 then?`
`10:08.41`	`Chris-NB`	`local prefix for salzburg`
`10:08.49`	`Chris-NB`	`0662`
`10:08.54`	`Chris-NB`	`or 0043662`
`10:09.08`	`Chris-NB`	`and for local calls I only need to dial 1234`
`10:09.14`	`Chris-NB`	`and not 06621234`
`10:10.24`	`JT`	`hrm`
`10:10.30`	`JT`	`so don't you dial international with 00?`
`10:11.48`	`*** join/#asterisk Growly (n=himself@125-238-7-142.broadband-telecom.global-gateway.net.nz)`
`10:12.28`	`Chris-NB`	`to make enum lookups I only need 436621234 so I strip the 00 away, make enum lookup. If I get no result, I need to check if it's a local, national or international nr, metch acording and dial the right nr`
`10:12.44`	`x86`	`http://www.wagenschenke.ch/HomeRun.swf`
`10:12.46`	`x86`	`lol`
`10:12.52`	`Chris-NB`	`*hrhr`
`10:12.57`	`Chris-NB`	`old, but funny : D`
`10:13.06`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`10:13.06`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`10:13.14`	`Makenshi`	`Chris-NB, i have a dial plan that does exactly that, would you like to see? it should be trivial to modify it to your requirements`
`10:13.34`	`Chris-NB`	`Makenshi, would be nice.`
`10:13.35`	`JT`	`Chris-NB: i don't know why you strip away 00, wouldn't it be easier to dial with it in place?`
`10:13.57`	`Makenshi`	`ok one sec lemme extract it onto pastebin`
`10:14.37`	`Chris-NB`	`JT, but for enum lookup the nr has to be in that format: <intern. prefix><national prefix><nr><extension> without leading 00 or 0`
`10:14.57`	`JT`	`what do you mean enum lookup?`
`10:15.22`	`Chris-NB`	`JT, look the number in the enum up, if there is a sip entry for this nr.`
`10:15.44`	`JT`	`do you have different providers for different countries?`
`10:15.49`	`Chris-NB`	`JT, enum - Electronic Numbering Mapping`
`10:15.58`	`JT`	`i know what enum is`
`10:16.00`	`Chris-NB`	`ok`
`10:16.13`	`JT`	`i don't know why you'd be doing lookups from the dialplan`
`10:16.21`	`JT`	`unless you were trying to be the next vonage :P`
`10:16.50`	`Chris-NB`	`I look if the call can be established via SIP/internet directly`
`10:16.55`	`Chris-NB`	`to save costs.`
`10:16.58`	`JT`	`well`
`10:17.01`	`Chris-NB`	`If not, I dial via pstn`
`10:17.10`	`Makenshi`	`http://pastebin.ca/327878`
`10:17.16`	`JT`	`isn't it pretty much the case that if the call is not going to your country, it goes via sip?`
`10:17.20`	`Makenshi`	`my plan takes into account the dialing codes`
`10:17.45`	`Makenshi`	`and uses enum also`
`10:17.50`	`phearless`	`How could I change the timeout before dialing a number on my Linksys/Sipura 942 ?`
`10:18.41`	`phearless`	`it seems to be 3s on my phone`
`10:18.44`	`phearless`	`by default`
`10:19.00`	`Chris-NB`	`thanks Makenshi. looks very good. and very well structured!`
`10:19.00`	`JT`	`what is the point in using enum unless you have multiple international routes, and very complicated ones at that?`
`10:19.11`	`Makenshi`	`Chris-NB, why thank you! i tried hard to make it that way :)`
`10:19.31`	`Chris-NB`	`Makenshi, cause u showed it to me.`
`10:20.24`	`JT`	`i don't know why you don't just do`
`10:20.25`	`Chris-NB`	`JT, If there is an enum entry for a given nr, you can call that nr directly via SIP and don't need to use pstn or a sip provider`
`10:20.26`	`*** join/#asterisk L\|NUX (n=linux@unaffiliated/lnux/x-10290)`
`10:20.32`	`JT`	`_00X.`
`10:20.38`	`Makenshi`	`JT, it saves money, for instance, let's me call toll-free numbers in a lot of countries for free`
`10:20.45`	`JT`	`_0662X.`
`10:20.55`	`Makenshi`	`eg, +18005558355`
`10:21.11`	`JT`	`so 1800 numbers, that's about it?`
`10:21.27`	`Makenshi`	`no, that was just an example`
`10:21.55`	`Makenshi`	`my number is in there, +441213146461`
`10:22.03`	`Makenshi`	`so you could call me free direct via sip`
`10:22.22`	`JT`	`does it use e164.org or something else?`
`10:22.40`	`Makenshi`	`e164.org`
`10:22.46`	`JT`	`ahh right`
`10:22.48`	`Makenshi`	`i've also registered the number range for my organization`
`10:22.49`	`JT`	`now i get it`
`10:22.51`	`Makenshi`	`1000 numbers`
`10:22.59`	`Chris-NB`	`JT, if you do a dig NAPTR 5.5.3.8.5.5.5.0.0.8.1.e164.org you get an SIP uri back`
`10:23.12`	`JT`	`it's that unofficial e164 setup by evilbunny`
`10:23.28`	`JT`	`i hate that dude`
`10:23.31`	`*** join/#asterisk geertn___ (n=geert@frantix.noc.ams-ix.net)`
`10:23.31`	`Makenshi`	`not many countries use e164.arpa yet unfortunately`
`10:23.33`	`JT`	`he lives in my city, heh`
`10:23.33`	`Makenshi`	`how come?`
`10:23.39`	`JT`	`he;s a complete idiot`
`10:23.50`	`Chris-NB`	`so I can call that nr directly via sip if I dial +18005558355 from my phone in austria`
`10:23.56`	`Chris-NB`	`so I save money : D`
`10:23.59`	`Makenshi`	`have a look at the records in 1.6.4.6.4.1.3.1.2.1.4.4.e164.org. :o)`
`10:24.09`	`Makenshi`	`there's not just sip`
`10:24.11`	`JT`	`arogant fool who mostly copies ideas from others`
`10:24.12`	`JT`	`heh`
`10:24.53`	`JT`	`sounds like one of his few projects that haven't died in the arse`
`10:25.05`	`Chris-NB`	`hey Makenshi. way to much entries : D`
`10:25.13`	`Chris-NB`	`I've only my sip entry : D`
`10:25.17`	`J4k3`	`so much anger`
`10:25.30`	`Makenshi`	`hehe`
`10:25.59`	`Chris-NB`	`but mine is in e164.arpa : D`
`10:26.06`	`Makenshi`	`how did you manage that?`
`10:26.08`	`JT`	`i told him his wireless plans were ridiculous and wouldn't work, and why`
`10:26.12`	`JT`	`heaps of people fanboyed him`
`10:26.19`	`JT`	`then 2 years later, i was proven right`
`10:26.28`	`J4k3`	`what was he trying to do?`
`10:26.33`	`J4k3`	`lemmie guess... mesh the planet!`
`10:26.35`	`Chris-NB`	`in austria e164.arpa is very well supported`
`10:26.47`	`Chris-NB`	`enum.at <-- you can register e164.arpa entries`
`10:26.56`	`Chris-NB`	`even delegate the entries to you (own dns)`
`10:27.03`	`Makenshi`	`wireless is a pita.. gonna be spending 16k this year on new wireless switches`
`10:27.07`	`JT`	`J4k3: community wireless in 2002 with a subscription fee system`
`10:27.13`	`Makenshi`	`Ahh cool, my country is really slow about such things`
`10:27.24`	`JT`	`"wpop"s only costing about $2500`
`10:27.34`	`JT`	`they were going to have 250GB HDDs, this is in 2002`
`10:27.38`	`JT`	`stupid idea`
`10:27.41`	`Makenshi`	`we have 80 waps over 2 campuses`
`10:27.45`	`J4k3`	`JT: haha... thats just silly.`
`10:27.47`	`JT`	`it never got off the ground`
`10:28.03`	`Makenshi`	`i hope wimax will crush wifi`
`10:28.08`	`J4k3`	`it won't.`
`10:28.17`	`JT`	`who wants to subscribe to what's meant to be a community project, and it doesn;'t come with Internet`
`10:28.25`	`Makenshi`	`can't say that, you never know :)`
`10:28.36`	`J4k3`	`well wifi's main thing is price at this point`
`10:28.41`	`J4k3`	`wimax radios aren't cheap`
`10:28.48`	`J4k3`	`they also, so far, don't perform amazingly well.`
`10:29.12`	`Makenshi`	`annoyingly there are 2 deployments in towns next to mine, but not here :/`
`10:29.26`	`J4k3`	`but most of the stuff on the market is designed for 2ghz+ spectrum, which means the laws of physics apply a lot harder than any protocol can compensate for`
`10:29.55`	`JT`	`so now there's almost no community wireless action in sydney because everyone got behind his idea, but was disappointed when it all went nowhere, and he eventually jumped ship`
`10:30.04`	`*** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290)`
`10:30.13`	`Makenshi`	`i just noticed a flaw in my dial plan.. it doesnt do enum lookups for local numbers, oops`
`10:30.31`	`JT`	`precious cents being wasted ;)`
`10:30.36`	`Makenshi`	`i am hoping to move back to sydney by the end of next year`
`10:30.41`	`Makenshi`	`i didn't know that's where he was`
`10:31.07`	`Makenshi`	`oh, scratch that comment, it does to enum lookups for local numbers, nvm`
`10:31.52`	`Makenshi`	`hey why not :)`
`10:31.59`	`J4k3`	`its free`
`10:32.02`	`J4k3`	`toll free calls too`
`10:32.03`	`Makenshi`	`exactly`
`10:32.06`	`J4k3`	`I've gotta have the line for backup`
`10:32.27`	`J4k3`	`its incoming calls will be forwarded to my did (unmetered incoming)`
`10:32.34`	`J4k3`	`unluckily we have no number portability here yet`
`10:32.34`	`JT`	`Makenshi: you from sydney?`
`10:33.25`	`Chris-NB`	`Makenshi, your enum lookups. do you really need these? e164.info-enum.org-e164.televolution.net`
`10:33.39`	`Chris-NB`	`Makenshi, cause you do 5 enum lookups for every call?`
`10:33.56`	`Makenshi`	`Chris-NB, if you don't want them you can easily take them out`
`10:34.02`	`Makenshi`	`i like to exhaust every possible alternative first`
`10:34.05`	`JT`	`so if enum is down or slow, calls are slow or don't happen? :)`
`10:34.07`	`Chris-NB`	`ok`
`10:34.23`	`Makenshi`	`JT, I lived there for a couple years 1999-2001 and gained permanent residency`
`10:34.33`	`Makenshi`	`england is a dump and i want to leave`
`10:34.52`	`JT`	`ah ok`
`10:35.02`	`JT`	`i want to visit england sometime :P`
`10:35.11`	`Makenshi`	`i lived in sa and wa for a little while too`
`10:35.24`	`JT`	`nice and hot`
`10:35.27`	`Chris-NB`	`JT, I've a local dns cache on the server, so if nothing is found, or its down, it doesn't take long to recognize`
`10:35.36`	`Makenshi`	`mmmm that's another thing, the weather here sucks`
`10:35.43`	`Chris-NB`	`I studied in stafford for one semester`
`10:35.45`	`JT`	`i hate hot weather`
`10:35.52`	`Makenshi`	`i love hot weather!`
`10:36.01`	`Chris-NB`	`and jep, weather in the uk sucks!`
`10:36.36`	`JT`	`hot weather makes me angry`
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`10:46.44`	`*** join/#asterisk amer (n=ame@202.125.139.246)`
`10:46.49`	`amer`	`hi`
`10:46.55`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`10:46.55`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`10:47.13`	`amer`	`is there any access list for asterisk, like we have for ssh?`
`10:49.06`	`Makenshi`	`hosts.allow/hosts.deny?`
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`11:20.21`	`nilkanthp`	`amer, what is the goal?`
`11:21.02`	`Makenshi`	`he gone`
`11:24.25`	`nilkanthp`	`Makenshi, have you done fax on asterisk?`
`11:24.51`	`Makenshi`	`nilkanthp, nope`
`11:25.22`	`nilkanthp`	`ok :)`
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`11:45.17`	`ThoMe`	`Hello`
`11:45.22`	`ThoMe`	`JT: Hey. :-)`
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`11:48.50`	`*** join/#asterisk my007ms (n=my007ms@217.139.224.194)`
`11:49.13`	`my007ms`	`hello`
`11:49.20`	`my007ms`	`i have somethng upnormal`
`11:49.48`	`my007ms`	`i use cisco phone 7940 with my asterisk`
`11:49.56`	`my007ms`	`it was work fine`
`11:50.27`	`my007ms`	`the i start get busy while i am sure not one use phone in the othere end of the line`
`11:52.17`	`Jenna`	`hi guys, Anyone with experience of using sangoma cards especailly http://www.sangoma.com/datasheets/p_a200-specs the analog telephony card with asterisk ?`
`11:54.04`	`*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)`
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`12:00.44`	`signius`	`.`
`12:04.20`	`*** part/#asterisk Bazy (n=bazy@89.137.178.124)`
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`12:14.47`	`zeeesh`	`hui`
`12:14.48`	`zeeesh`	`hi`
`12:18.40`	`bcnl`	`!help`
`12:18.53`	`bcnl`	`damn, I swear I saw a bot in here previously`
`12:21.16`	`*** part/#asterisk joelsolanki (i=joelsola@202.160.161.94)`
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`12:28.51`	`zeeesh`	`if i want to dial ... how can i dial .. access number through xpro ... is this extensions is right or not ? " exten => XXXXXX,1,Dial(sipserver/XXXX@XXXXX) " ???`
`12:30.16`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`12:32.43`	`*** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk)`
`12:33.47`	`zoa`	`hey ho!`
`12:34.26`	`anonymouz666`	`let's go!`
`12:39.19`	`zeeesh`	`what does it mean.."" == Parsing '/etc/asterisk/sip_notify.conf': Found`
`12:39.19`	`zeeesh`	`<PROTECTED>`
`12:39.39`	`Chris-NB`	`that the file /etc/asterisk/sip_notify.conf was found`
`12:39.48`	`Chris-NB`	`and got parsed`
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`12:41.18`	`zeeesh`	`ok`
`12:41.29`	`zeeesh`	`so what about this if i want to dial ... how can i dial .. access number through xpro ... is this extensions is right or not ? " exten => XXXXXX,1,Dial(sipserver/XXXX@XXXXX) " ???`
`12:42.28`	`Chris-NB`	`is XXXXXX a number or do you want to match for numbers?`
`12:42.38`	`Chris-NB`	`if you want to match, you have to write _XXXXXX`
`12:43.50`	`zeeesh`	`like`
`12:44.30`	`Chris-NB`	`_XXXXXX matches numbers 000000 - 999999`
`12:45.20`	`Chris-NB`	`otherwise you have to write the number exten => 123456,1,Dial(SIP/123456@sipprovider)`
`12:45.26`	`zeeesh`	`if this is my access number what i want to dial through using ... xpro ... " exten => 22222,1,Dial(sipserver/22222@22222)???? will it or something else`
`12:45.53`	`Chris-NB`	`should this bee a SIP-call?`
`12:46.34`	`Chris-NB`	`then the Dial should be Dial(SIP/sipprovider/22222)`
`12:46.49`	`zeeesh`	`ok`
`12:46.55`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`12:46.55`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`12:47.41`	`Chris-NB`	`works if you have a [sipserver] in your sip.conf`
`12:47.42`	`Chris-NB`	`with username and secret in it`
`12:48.54`	`Chris-NB`	`or Dial(SIP/22222@sipserver)`
`12:50.22`	`*** join/#asterisk uwe (n=uwe@dogbert.palnet.com)`
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`12:54.40`	`my007ms`	`i need to remove one line of sip show peers`
`12:54.48`	`my007ms`	`without restart my asterisk`
`12:56.50`	`clorabit`	`JT: are u there ?`
`12:56.57`	`zeeesh`	`what is the difference about .. " username " and the " fromuser"????`
`13:03.44`	`clorabit`	`hello anyone can help me with my asterisk configuration`
`13:03.46`	`my007ms`	`<PROTECTED>`
`13:03.48`	`my007ms`	`<PROTECTED>`
`13:04.35`	`clorabit`	`i've install it but when i dial to echo / playback extension there are no sound at all`
`13:05.19`	`clorabit`	`log showing that i asterisk receive connection but no sound at client`
`13:05.21`	`*** join/#asterisk Aurs (n=Aurs@cBD7BBF51.dhcp.bluecom.no)`
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`13:20.57`	`merbzt`	`when I'm using Dial(...,M()) the macro doesn't terminate if the one calling hangs up, is there anything that can change that ?`
`13:21.08`	`*** join/#asterisk dimmik (n=root@ios.intranet.GR)`
`13:25.48`	`zeeesh`	`oye gandooo ..jhooot bolna hai`
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`14:04.21`	`Dr-Linux`	`what does this warning mean: Jan 25 06:02:28 WARNING[25129]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 129 scheduled tasks all at once`
`14:09.32`	`Makenshi`	`Yesterday Siemens received the biggest fine ever given by the EU for running a cartel`
`14:09.54`	`Gido-E`	`Dr-Linux exact what it says.`
`14:10.20`	`Dr-Linux`	`Gido-E: i'm sorry but what didn't understand`
`14:11.02`	`coppice`	`Makenshi: I wonder how they will pay that, when they can't seem to make any money :-)`
`14:11.04`	`*** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net)`
`14:11.28`	`Makenshi`	`coppice, good question..`
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`14:25.09`	`*** mode/#asterisk [+o anthm] by ChanServ`
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`14:30.52`	`earthsound`	`has mark been in here the past few days?`
`14:31.29`	`yassine`	`neither me nor the caller can hear each other and when i dial a wrong nummber i can here the operator tell me that the person call is not available ... any idea why i can hear the operator and not the caller ?`
`14:34.49`	`SomeOne1`	`can someone make a kickass professional looking flash intro for me?`
`14:34.53`	`SomeOne1`	`i'll pay $50 through paypal`
`14:34.57`	`SomeOne1`	`i'm desperate for it, and im good on the payment thing`
`14:35.00`	`SomeOne1`	`but i need it done RIGHT now`
`14:36.10`	`*** join/#asterisk reber (i=reber@gateway/tor/x-b5d890ffc31b5dec)`
`14:36.18`	`yassine`	`SomeOne1, why dont you try #flash`
`14:36.46`	`earthsound`	`another scott adams reference to linux: http://www.dilbert.com/comics/dilbert/archive/dilbert-20070125.html`
`14:39.14`	`*** part/#asterisk zerowarez (n=zero@201.29.209.42)`
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`14:39.39`	`Faithful`	`earthsound: you have to be a linux geek to think that's funny`
`14:39.47`	`Crescendo`	`What ports need to be forwarded to the server in order for a WAN Cisco IP phone to work through NAT?`
`14:41.38`	`*** join/#asterisk nilkanthp (n=nilkanth@202.189.249.206)`
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`14:44.27`	`tsurko`	`hello`
`14:45.18`	`tsurko`	`I've a problem with asterisk-gui. I've modified manager.conf and http.conf, reload asterisk's configuration, but nothing answers on the specified port`
`14:45.37`	`tsurko`	`Do you have any ideas where the problem is?`
`14:45.44`	`mercestes`	`tsurko: Ooo! I do! I do!`
`14:46.10`	`mercestes`	`tsurko: Your problem is....shakes magic 8 ball Your asking #asterisk-gui questions in #asterisk instead of #asterisk-gui.`
`14:46.18`	`tsurko`	`ups`
`14:46.22`	`tsurko`	`I do apologise!`
`14:46.27`	`mercestes`	`hehe...it's ok.`
`14:46.34`	`mercestes`	`I appreciate every opportunity..:)`
`14:46.41`	`tsurko`	`just noticed it in the topic`
`14:46.50`	`mercestes`	`lol. It's comic relief at this point..`
`14:46.55`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`14:46.55`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`14:47.03`	`mercestes`	`but yea, They'd be able to help you. be sure to mention your OS btw.`
`14:47.14`	`tsurko`	`okay thank you a lot`
`14:47.30`	`mercestes`	`I'm guessing it's something to do with your apache setup and your "listen" variable in httpd.conf in /etc/apache`
`14:47.39`	`tzanger`	`sweet`
`14:47.40`	`mercestes`	`but I honestly have no clue beyond that`
`14:47.45`	`tzanger`	`twinkle seems like a pretty decent little linux qt sip phone`
`14:48.52`	`tsurko`	`mercestes, i've checked that, but whatever... I've spammed enough for today:)`
`14:51.10`	`Crescendo`	`Asterisk should support UPnP. :)`
`14:52.06`	`yatesy`	`for what purpose?`
`14:54.20`	`*** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net)`
`14:57.36`	`*** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg)`
`14:57.40`	`skirmisha`	`hello all`
`14:58.17`	`skirmisha`	`does anyone know if latest svn version of asterisk support RTP packetization`
`14:59.30`	`skirmisha`	`?`
`14:59.43`	`doolph`	`what is rtp packetization`
`15:00.14`	`skirmisha`	`To set a desired packetization interval on a specific codec,`
`15:00.14`	`skirmisha`	`append that inteval to the allow= statement.`
`15:00.14`	`skirmisha`	`Example:`
`15:00.14`	`skirmisha`	`allow=ulaw:30,alaw,g729:60`
`15:01.30`	`doolph`	`for what is that`
`15:01.48`	`*** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net)`
`15:01.53`	`skirmisha`	`huh`
`15:02.10`	`elriah`	`Hi all. If I'm trying to join one asterisk box to another via sip, do I neeed a register command and a peer definition on BOTH boxes in sip.conf?`
`15:02.53`	`doolph`	`elriah, if you want to call each other yes`
`15:03.05`	`Gido-E`	`elriah use IAX`
`15:03.19`	`elriah`	`If I want box "A" to just call box "B", I need a register on box "A" and a peer definition on box "B", right?`
`15:03.27`	`elriah`	`Gido-E: It's not an option in this config.`
`15:03.40`	`elriah`	`Gido-E: i.e., I don't make the remote box's policy.`
`15:04.38`	`Crescendo`	`What ports need to be routed to the server to use a Cisco IP phone outside?`
`15:04.42`	`elriah`	`If I want box "A" to just call box "B", I need a register on box "A" and a peer definition on box "B", right?`
`15:04.48`	`elriah`	`Or do I have that backwards?`
`15:07.50`	`*** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net)`
`15:11.20`	`phearless`	`How could I change the timeout before dialing a number on my Linksys/Sipura 942 ?`
`15:12.42`	`Chris-NB`	`Crescendo, TCP Port 5060`
`15:13.25`	`Crescendo`	`Chris-NB, already done, still not receiving audio on the server side.`
`15:13.30`	`Chris-NB`	`Crescendo, my experience is that the udp ports 20k-30k are opened when the 1st packets travel from asterisk outside`
`15:13.47`	`Chris-NB`	`Crescendo, made this with sonicwall and checkpoint`
`15:14.10`	`Chris-NB`	`Crescendo, rtp packets use a rand port between 20k -30k`
`15:14.27`	`Chris-NB`	`Crescendo, deffined by asterisk. but you can set that range`
`15:16.55`	`elriah`	`If I want box "A" to just call box "B", I need a register on box "A" and a peer definition on box "B", right?`
`15:17.50`	`*** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman)`
`15:20.16`	`skirmisha`	`anyone who can tell me this`
`15:20.21`	`skirmisha`	`does anyone know if latest svn version of asterisk support RTP packetization`
`15:23.36`	`Chris-NB`	`what is packetization?`
`15:23.42`	`file`	`of what? there's 1.2, 1.4, and trunk`
`15:23.45`	`file`	`1.4 and trunk support it`
`15:24.31`	`skirmisha`	`file svn`
`15:24.51`	`skirmisha`	`i am not sure what standart svn downloads`
`15:25.12`	`file`	`SVN is a version control system, it's where all the Asterisk source code is kept for all the different versions...`
`15:25.13`	`*** join/#asterisk CPSK (n=CPSK@c6.ars.ba.nextra.sk)`
`15:25.16`	`file`	`so saying SVN could mean any of them`
`15:25.31`	`*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)`
`15:25.37`	`*** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue)`
`15:25.50`	`*** join/#asterisk bagan_jermal (n=mfaizaly@205.60.95.219.cbj02-home.tm.net.my)`
`15:26.12`	`Crescendo`	`Where are the RTP ports configured?`
`15:26.22`	`yatesy`	`rtp.conf`
`15:26.30`	`Crescendo`	`Lol, oops`
`15:26.32`	`Crescendo`	`Didn't see it.`
`15:27.17`	`skirmisha`	`file i need trunk right?`
`15:27.24`	`Crescendo`	`In the scenario with Cisco IP phones, should I increase the upper RTP end to 32766?`
`15:27.32`	`file`	`if you are asking that question then no`
`15:27.39`	`file`	`1.2 does not have the RTP packetization support, 1.4 does`
`15:28.11`	`mercestes`	`omg...not this jerk again`
`15:28.20`	`*** join/#asterisk af_ (n=af@ip-173-157.sn1.eutelia.it)`
`15:28.42`	`*** join/#asterisk svenna_ (n=svenna@p548D26E3.dip0.t-ipconnect.de)`
`15:28.55`	`sevard`	`no, fucking, way.`
`15:29.13`	`skirmisha`	`file what about the patch released`
`15:29.23`	`clorabit`	`hello anyone can help my problem`
`15:29.24`	`phearless`	`exten => i,1,Answer`
`15:29.24`	`phearless`	`exten => i,2,Playback(pbx-invalid)`
`15:29.24`	`phearless`	`exten => i,3,Hangup`
`15:29.30`	`phearless`	`it should work, right?`
`15:29.37`	`skirmisha`	`file bug id 5162`
`15:29.52`	`clorabit`	`i've post it here http://forums.digium.com/viewtopic.php?t=13097`
`15:29.58`	`file`	`skirmisha: then you can use the patch if it works`
`15:30.18`	`mercestes`	`clorabit: should work. What is you rproblem?`
`15:30.52`	`clorabit`	`mercestes: no playback sound`
`15:31.00`	`elriah`	`Guys, if I'm making a call from one asterisk box to another, and I can't use IAX only SIP, on which box do I put the peer definition and on which box to I put the register definition and how do I format the dial command? Thanks for any help. I'm struggling here. IAX is easy, this has confused me.`
`15:31.16`	`skirmisha`	`ok`
`15:31.50`	`sevard`	`elriah: it's basically the same as IAX`
`15:32.23`	`sevard`	`elriah: I believe there's an article on voip-info.org; why can't you use iax (curious)?`
`15:33.09`	`elriah`	`sevard: the remote's policy won't allow for IAX (paper policy, i.e., business decision).`
`15:33.53`	`sevard`	`strange, did you try to convince them that connecting asterisk to asterisk with IAX is a much better decision?`
`15:33.57`	`*** join/#asterisk phsultan (n=phsultan@PO-47165.rocqadm.inria.fr)`
`15:34.09`	`elriah`	`lol, yea.`
`15:35.22`	`sevard`	`I also believe there's an example of registration in sip.conf.sample`
`15:35.26`	`sevard`	`check that out, i'm off to school.`
`15:35.31`	`sevard`	`good luck mate.`
`15:35.53`	`elriah`	`Thanks ;)`
`15:36.41`	`*** join/#asterisk FaithX (n=faithful@ns.linuxterminal.com)`
`15:36.46`	`clorabit`	`mercestes: i've this situation comp A with config user 1234 and comp B with 1235, when A dial to B, B can ring2 but not for B dial to A or perhaps there are missing configuration`
`15:37.15`	`clorabit`	`anyone can solve my problem ?`
`15:37.20`	`*** join/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net)`
`15:37.29`	`mercestes`	`clorabit: blame one way audio and go from there.`
`15:37.54`	`clorabit`	`mercestes: what u mean with one way audio ?`
`15:38.12`	`mercestes`	`... i could explain it but I would come across as condescending and sarcastic.`
`15:38.30`	`mercestes`	`it literally means "one way audio" as in audio is only going one way.`
`15:38.46`	`mercestes`	`which is usually caused by NAt, firewall issues, or port issues.`
`15:39.02`	`clorabit`	`ic ...`
`15:39.08`	`drray`	`anyone familar with red alarms on adit 600's?`
`15:39.08`	`sevard`	`elriah: if you can't figure it out, here's a reference (ignore all the freepbx shit and iax stuff) http://forum.voxilla.com/asterisk-users-group/no-sound-sip-16161.html`
`15:39.40`	`elriah`	`Thanks! :)`
`15:40.38`	`*** join/#asterisk XIN01OZ (n=KNOCK@c-68-63-34-189.hsd1.al.comcast.net)`
`15:42.26`	`clorabit`	`mercestes: i've turn off all firewall. can it become a problem when comp i use for asterisk also become a NAT gateway ?`
`15:42.50`	`mercestes`	`clorabit: first queston.....is this all on the same box or two different boxes?`
`15:43.29`	`clorabit`	`asterisk comp and nat gateway is a same box`
`15:43.51`	`clorabit`	`client run on 2 other difference box`
`15:46.52`	`phearless`	`hello guys!`
`15:46.55`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`15:46.55`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`15:47.14`	`phearless`	`exten => i,1,Answer`
`15:47.14`	`phearless`	`exten => i,2,Playback(pbx-invalid)`
`15:47.14`	`phearless`	`exten => i,3,Hangup`
`15:47.17`	`phearless`	`is it right?`
`15:47.32`	`phearless`	`it is in my [default] of extensions.conf`
`15:47.39`	`mercestes`	`clorabit: ...`
`15:47.55`	`phearless`	`but when I dial a wrogn extension, I got no error message`
`15:47.59`	`mercestes`	`clorabit: You have a phone....and Asterisk. Is the phone and asterisk on the same box?`
`15:48.24`	`clorabit`	`mercestes: i think i found some wrong in my box configuration here ..`
`15:48.42`	`clorabit`	`brb changing ip ...`
`15:48.46`	`mercestes`	`nods`
`15:48.47`	`mercestes`	`Good luck`
`15:49.55`	`jojo^`	`Suppose I were to set up a sales office with about 15 employees. What IAX-based softphones would you recommend? I am also yet to decide between Linux-based computers or Windows-based.. The quality of softphone could make me favour in one or the other direction. I've only tested diax on Windows so far, but I've seen there is kiax, and maybe others too?`
`15:50.37`	`clorabit`	`mercestes: works!! :)`
`15:50.59`	`clorabit`	`mercestes: thanks bro`
`15:51.01`	`elriah`	`If I want to do a one-way asterisk to asterisk call with iax2, I only have to define a peer in the RECEIVING hosts iax2.conf then just use a dial() cmd on the CALLING host to make the call, right?`
`15:51.05`	`*** join/#asterisk asnowden (n=dont@196.7.14.163)`
`15:52.14`	`phearless`	`mercestes clorabit , any clue ?`
`15:52.34`	`clorabit`	`mercestes: my problem is cause by i'm access from different ip address which 1234 extension is access from 192.x.x.x subnet and 1235 exten connect from 10.x.x.x subnet`
`15:53.34`	`clorabit`	`mercestes: my asterisk box is working as a router and have 3 difference subnet perhaps this cause the problem :)`
`15:53.42`	`asnowden`	`hey guys. i'm trying to get asterisk working with multiple frame ABR speex, i've hacked the respective functions in codec_speex.c and from what I can see it is generating the correct data into the ast_frame, but by the time the rtp packet gets written there are a seemingly random number of "random" characters that have been inserted before my payload data. Anyone have any idea why or where this happens?`
`15:54.27`	`clorabit`	`phearless: i'm a newbie but if you don't mind can u explain your problem perhaps we can discuss together`
`15:54.47`	`phearless`	`ok clorabit`
`15:54.48`	`phearless`	`<phearless> exten => i,1,Answer`
`15:54.48`	`phearless`	`<phearless> exten => i,2,Playback(pbx-invalid)`
`15:54.48`	`phearless`	`<phearless> exten => i,3,Hangup`
`15:54.54`	`phearless`	`it is in my [default] of extensions.conf`
`15:55.08`	`phearless`	`but I do not hear the message if I dial a wrong exten`
`15:55.26`	`phearless`	`I do not know if there is a special order of the lines in extensions.conf...`
`15:55.46`	`clorabit`	`wait let me try those config in my box`
`15:55.57`	`phearless`	`ok`
`15:56.28`	`phearless`	`I use Asterisk 1.2.10`
`15:57.02`	`*** join/#asterisk voipgeek (n=support@206.mui23.chcg.cgcil02r18.dsl.att.net)`
`15:57.39`	`clorabit`	`have u check log message ?`
`15:58.26`	`voipgeek`	`Can anyone recommend a place to choose/purchase DID numbers? I want to pay by the minute, and terminate it myself with SIP. I'm looking for numbers in the US (Chicago/Madison).`
`15:59.05`	`clorabit`	`phearless: those not work in my box also, i'm using 1.2.14`
`15:59.47`	`phearless`	`<clorabit> have u check log message ? <--- yes, nothing interesting`
`15:59.48`	`mercestes`	`clorabit: NAT issue then..:) Your welcome.`
`16:00.20`	`elriah`	`On an outbound IAX2 call, how do I specify the context on the RECEIVING asterisk server?`
`16:01.35`	`clorabit`	`phearless: is your log file also look like this "Jan 25 22:57:18 NOTICE[4817]: chan_iax2.c:6931 socket_read: Rejected connect attempt from 10.2.100.100, request '12221@default' does not exist"`
`16:01.38`	`*** join/#asterisk saftsack (n=saftsack@p54A7DD3E.dip.t-dialin.net)`
`16:02.32`	`elriah`	`On an outbound IAX2 call, how do I specify the context on the RECEIVING asterisk server?`
`16:03.06`	`phearless`	`<PROTECTED>`
`16:03.06`	`phearless`	`Jan 25 16:00:28 WARNING[23781]: chan_sip.c:1980 create_addr: No such host: 522`
`16:03.06`	`phearless`	`Jan 25 16:00:28 NOTICE[23781]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)`
`16:03.06`	`phearless`	`<PROTECTED>`
`16:03.06`	`phearless`	`<PROTECTED>`
`16:03.08`	`phearless`	`Jan 25 16:00:28 WARNING[23781]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '522'`
`16:03.12`	`phearless`	`I got this!`
`16:06.36`	`*** join/#asterisk marv[work] (n=timr@24.214.206.254)`
`16:06.43`	`mquin`	`elriah: IAX2/user@host/extension@context`
`16:06.58`	`elriah`	`Thanks.`
`16:09.51`	`*** join/#asterisk Overworked554 (n=Overwork@12-226-108-103.client.mchsi.com)`
`16:10.10`	`*** join/#asterisk svenna_ (n=svenna@p548D26E3.dip0.t-ipconnect.de)`
`16:11.21`	`elriah`	`mquin: If there is no extension, you just want to drop in a context, just omit the "extension@", right?`
`16:11.40`	`*** join/#asterisk Chris-NB (n=chris@88.117.140.228)`
`16:11.54`	`*** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir)`
`16:12.44`	`*** join/#asterisk jm\|home (n=jm\|home@dilbert.jamiem.com)`
`16:12.53`	`mquin`	`elriah: possibly, or use 's'`
`16:13.44`	`elriah`	`Thanks. I keep getting: Rejected connect attempt from 72.149.159.16, who was trying to reach 's@whatevercontext'`
`16:15.07`	`clorabit`	`phearless: try to use _. instead i extension`
`16:15.19`	`clorabit`	`works for me ..`
`16:15.47`	`clorabit`	`phearless: but still don't know why it's not work with i`
`16:16.49`	`phearless`	`I just tried http://www.planetwayne.com/forums/viewtopic.php?t=218`
`16:16.52`	`phearless`	`it seems to work`
`16:17.09`	`phearless`	`i seems to be impossible to use for dialled numbers`
`16:17.30`	`mquin`	`elriah: do you have an 's' (i.e. defautlt) extension defined in the context you are using?`
`16:17.33`	`*** join/#asterisk hohum (n=dcorbe@mercury.sunrocket.com)`
`16:18.20`	`wunderkin`	`umm _. picks up i, h, t, everything, not good... and in that case i will not work`
`16:18.41`	`elriah`	`Yep... Weird... my dial string is: dial(IAX2/my.remote.asterisk.host/s@somecontext,90,r)`
`16:19.37`	`wunderkin`	`you can get rid of r too`
`16:20.12`	`*** join/#asterisk Nugget (i=nugget@dazed.notslacker.com)`
`16:20.13`	`elriah`	`Would the rejected attempt be auth related or something to do with the context?`
`16:20.48`	`wunderkin`	`maybe an iax2 debug will help but probably auth related`
`16:21.10`	`*** join/#asterisk Nukemizer (n=Nuke@67.137.28.165)`
`16:23.05`	`CrazyTux`	`Where can I find documentation on using *?`
`16:23.19`	`CrazyTux`	`I'm looking at voip-info wiki, but not much detail?`
`16:23.24`	`elriah`	`CAUSE: No authority found...`
`16:23.25`	`CrazyTux`	`I'm looking for specifically 1.4`
`16:23.27`	`elriah`	`hrm...`
`16:24.18`	`elriah`	`Do I have to have an iax.conf entry on the calling machine if I specify username/pass in the dial command?`
`16:26.59`	`*** part/#asterisk Overworked554 (n=Overwork@12-226-108-103.client.mchsi.com)`
`16:27.39`	`clorabit`	`phearless, mercestes gtg`
`16:28.57`	`mercestes`	`l8s`
`16:28.58`	`mercestes`	`...`
`16:28.58`	`mercestes`	`I want to club a baby seal`
`16:29.24`	`*** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla)`
`16:29.24`	`*** mode/#asterisk [+o russellb] by ChanServ`
`16:29.52`	`zoa`	`hey ho russie`
`16:29.53`	`wltjr`	`anyone here familiar with CallingPres/SetCallingPres ?`
`16:29.54`	`*** join/#asterisk Overworked554 (n=ken@12-226-108-103.client.mchsi.com)`
`16:30.25`	`*** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca)`
`16:30.26`	`wltjr`	`trying to figure out of I need to set that to a particular value to specify cid on outbound call over a pri`
`16:30.42`	`mercestes`	`wltjr: Don't see why you would have to.`
`16:30.48`	`CPSK`	`Hi, looking for help with Tenovis Integral E1 and TE110P.. msg me, thanx`
`16:30.57`	`wltjr`	`mercestes: everything I am reading seems to imply it's used for incoming calls?`
`16:31.14`	`mercestes`	`wltjr: IS this in zapata.conf or extensions.conf?`
`16:31.47`	`mercestes`	`CPSK: I know about 50 ppl who answered you yesterday when you spammed us forever and according to them, you never answered anyone.`
`16:31.53`	`wltjr`	`mercestes: well I turned it on in zapata.conf, "usecallingpres=yes", but then in extensions.conf where I am setting the cid, I was not sure if I needed to call that function or not`
`16:33.29`	`mercestes`	`wltjr: Shouldn't need it, should default to something usable`
`16:33.35`	`wltjr`	`mercestes: one doc said it should be called before placing an outgoing call`
`16:34.06`	`mercestes`	`wltjr: I pass callerID and I don't have a "CallingPres" anywhere.`
`16:34.09`	`wltjr`	`mercestes: just trying to figure out what I need to do, in order to specify CID on outbound calls via pri`
`16:34.15`	`wltjr`	`mercestes: ok sweet`
`16:34.26`	`mercestes`	`wltjr: It should be called before setting CallerID, if you are going to call it.`
`16:34.28`	`wltjr`	`mercestes: does that effect inbound cid info?`
`16:34.33`	`wltjr`	`ah`
`16:34.57`	`mercestes`	`You shouldn't be responsible for inbound CID tho you can reset it if you wish.`
`16:35.14`	`wltjr`	`mercestes: just want to make sure there is nothing special I need to do to receive it :)`
`16:35.35`	`wltjr`	`mercestes: whole point to the pri is to be able to specify cid on outbound, and capture it on inbound`
`16:35.41`	`rene-`	`guys, how come i can chanspy all my extensions but in one i just get beeps, and the extension is up in a call?`
`16:35.45`	`*** join/#asterisk tclark (n=TC@S0106000f66c5d294.gv.shawcable.net)`
`16:35.55`	`CrazyTux`	`How can I pass the incoming SIP uri as the voicemail mailbox to check?`
`16:36.13`	`wltjr`	`granted some calls might be private, blocked, hoping the cid info provided via 888 number will resolve that, since 888 must know # for billing purposes`
`16:36.13`	`mercestes`	`wltjr: NOthing really special about it that I've had to play with`
`16:36.14`	`elriah`	`rene-: It's the flux capacitor...`
`16:36.20`	`wltjr`	`mercestes: great, ty`
`16:36.29`	`tclark`	`what is the fav no brainer iax softphone to install these daz`
`16:36.31`	`*** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com)`
`16:36.36`	`CrazyTux`	`so say I send sip:VOICEMAIL_MAILBOX_TO_CHECK@whatever -> asterisk grabs VOICEMAIL_MAILBOX_TO_CHECK, and asks 'enter password'`
`16:36.42`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`16:36.43`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`16:37.04`	`rene-`	`elriah: i knew it`
`16:37.14`	`*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)`
`16:37.20`	`elriah`	`lol`
`16:37.21`	`rene-`	`i have been telling to those guys not to mess with it`
`16:37.36`	`ctooley`	`Looking for a full time Asterisk/Linux administrator in Austin TX. We've got a large number of servers handling a very large number of calls. If you're interested, email chris.tooley@excel.com`
`16:38.28`	`mquin`	`CrazyTux: ${EXTEN} will get you that bit`
`16:38.55`	`mquin`	`actually, I'm waffling`
`16:39.01`	`CrazyTux`	`mquin, so exten => _89XXXXXXXXXX,1,Voicemail(${EXTEM}) ?`
`16:39.11`	`CrazyTux`	`EXTEN*`
`16:40.49`	`mquin`	`I think you'd need to use ${CALLERIDNUM}`
`16:41.08`	`mquin`	`you'll probably need some sort of security arrangement to stop the calling party from just spoofing that, though`
`16:41.24`	`*** join/#asterisk BrokenNoze (n=Simon@62.253.194.107)`
`16:41.31`	`*** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net)`
`16:41.33`	`CrazyTux`	`mquin, what do you recommend?`
`16:41.50`	`CrazyTux`	`mquin, well they'd have to enter their password to check the voicemail...`
`16:41.56`	`BrokenNoze`	`Hi, is there a simple way to add a batch of SIP devices to a server without adding a single entry into SIP.conf for each one?`
`16:42.10`	`mquin`	`CrazyTux: no idea - I'm by no means an expert on *`
`16:42.19`	`mquin`	`just thinking out loud, mostly`
`16:42.46`	`ScottyTM`	`HI`
`16:42.48`	`*** join/#asterisk Marlow (n=marlow@office.imagine.ie)`
`16:43.01`	`ScottyTM`	`upps, sorry caps was on`
`16:43.07`	`raidenz`	`What's going on with svn (1.4)? Each time I check out the 1.4 branch it's stuck at revision "51363" for the last few days.`
`16:43.13`	`Marlow`	`hi boys :)`
`16:43.17`	`ScottyTM`	`what's the reason for "WARNING[6330]: db.c:67 dbinit: Unable to open Asterisk database"`
`16:43.49`	`raidenz`	`missing and possibly permissions issues?`
`16:45.00`	`ScottyTM`	`raidenz: sounds good`
`16:45.29`	`ScottyTM`	`ahh, /var/lib/asterisk belonged to root`
`16:45.54`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`16:47.53`	`*** join/#asterisk blebleble (i=godie@caesar.godie.net)`
`16:47.55`	`raidenz`	`:)`
`16:48.50`	`blebleble`	`Anyone have an idea why the ARI wouldn't be showing calls that were outbound and monitored? They are actually be monitored and i see see the .wav file in /var/spool/asterisk/monitor but they aren't listing in the ARI for that user (inbound works fine but no out)`
`16:52.58`	`tzafrir_laptop`	`anyboddy connecting to France Telecom with BRI?`
`16:54.07`	`Marlow`	`blebleble: sounds like it's identifying the records belonging to the destination, channel instead of pairing them with a agent/extension`
`16:55.10`	`*** join/#asterisk chiang_sg (i=kodok@bb121-6-186-250.singnet.com.sg)`
`16:55.31`	`chiang_sg`	`hi, my asterisk trow this "pbx_find_extension: No such switch 'IAX'`
`16:55.38`	`chiang_sg`	`what does it mean ?`
`16:56.33`	`blebleble`	`Marlow: yah in /var/spool/asterisk/monitor the outbound show as OUT99-20070125 to where the inbound show as 20070125-114339`
`16:57.00`	`mquin`	`chiang_sg: you've probably typed IAX somewhere when you meant IAX2`
`16:57.21`	`chiang_sg`	`oh`
`17:01.21`	`variable_office`	`is there any way to simulate a whole lotta' calls for throughput & pps testing? (for ulaw)`
`17:02.09`	`*** part/#asterisk blebleble (i=godie@caesar.godie.net)`
`17:03.21`	`tzafrir_laptop`	`chiang_sg, it needs to be IAX2`
`17:03.37`	`tzafrir_laptop`	`chiang_sg, using the asterisk GUI with 1.4.0?`
`17:05.53`	`*** join/#asterisk De_Mon (n=de_mon@fl-76-4-98-162.dhcp.embarqhsd.net)`
`17:06.37`	`De_Mon`	`how to do I 'goto' exten => h,n(hangup),Hangup()`
`17:07.07`	`*** part/#asterisk asnowden (n=dont@196.7.14.163)`
`17:07.15`	`De_Mon`	`exten => h,1,GotoIf($["1" = "0"]?:n(hangup)) didn't work`
`17:07.50`	`Strom_C`	`De_Mon: the h extension is only executed /after/ the channel hangs up`
`17:09.02`	`De_Mon`	`where would I send the channel to exit the hangup extension?`
`17:09.24`	`Strom_C`	`what are you trying to do?`
`17:10.40`	`De_Mon`	`I used the G dial option to put caller/callee into a meetme conference. When they both leave the callerID is screwed up for the caller, so I reset the callers callerid and just dump the other members via MeetMeAdmin's Kick everyone option.`
`17:11.04`	`*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)`
`17:11.52`	`De_Mon`	`so, set callers callerid in the h extension and kicke veryone from conference, and do nothing for all the other members that go through the same h extension`
`17:11.54`	`*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)`
`17:12.33`	`Strom_C`	`De_Mon: why are you putting both parties in a meetme?`
`17:13.17`	`De_Mon`	`because its a conference call where more than 2 members will be present`
`17:13.34`	`De_Mon`	`everyone else gets an email saying join conference suchandsuch`
`17:14.39`	`Strom_C`	`and why can't the called party just dial into the conference bridge himself?`
`17:15.35`	`De_Mon`	`what does that have to do with the h extension?`
`17:16.43`	`Strom_C`	`De_Mon: because your initial question is kind of bizarre, and I'm trying to narrow down whether you're experiencing a greater architectural problem overall`
`17:16.51`	`Strom_C`	`i.e. "are you making this too complicated?"`
`17:17.00`	`De_Mon`	`Yeah it is pretty weird.`
`17:18.20`	`De_Mon`	`Caller calls in and enters someones extension, it spawns a meetme room for the caller and the called party and notifies other members of the group via email someone has called in. other members join. If the called party hangs up (marked to kill conf when he leaves) everyone is kicked out`
`17:18.22`	`*** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk)`
`17:18.43`	`De_Mon`	`if caller leaves, I want everyone to get kicked out too.`
`17:18.56`	`elriah`	`@$&(^@(&^%@&(&^($% I've been fighting a IAX/SIP call issue for four hours to find out that our jackass sysadmin has jacked up our dns servers (inside vs outside) arrrgggghhhhh!!!!`
`17:19.07`	`De_Mon`	`if they are both marked as kill conf when these people leave they both have to leave to satisfy the condition`
`17:19.49`	`Strom_C`	`De_Mon: that's in the realm of "highly unorthodox" - what exactly is the reason for doing things that way?`
`17:20.02`	`De_Mon`	`I am being paid to :]`
`17:20.30`	`Strom_C`	`wrong answer`
`17:20.33`	`*** part/#asterisk Marlow (n=marlow@office.imagine.ie)`
`17:20.35`	`De_Mon`	`its in the that doesnt make sense but if you want to do it, I can realm`
`17:20.35`	`Strom_C`	`what exactly is the reason for doing things that way?`
`17:21.22`	`Strom_C`	`is this one of those "the client wants me to do this weirdo thing and i'm just going to implement it without asking questions first" deals?`
`17:21.35`	`chiang_sg`	`if i have switch => Iax2/a:b@1.2.3.4 in my extensions.conf how do i check that the server know other server dialplan`
`17:22.05`	`*** join/#asterisk yelmans (n=bitlord6@adsl196-114-245-217-196.adsl196-16.iam.net.ma)`
`17:22.11`	`*** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca)`
`17:22.41`	`*** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it)`
`17:23.07`	`yelmans`	`hi all ,please help ,i want i have asterisk 1.2 on debian kernel 2.6 i want to use my 3 internet connection to share traffic over them ,any suggestions?`
`17:23.18`	`De_Mon`	`no.. They want to know how long each member was in the conference, and if one side leaves and everyone forgets to hangup it skews their data`
`17:23.34`	`Strom_C`	`that's not what i was asking, De_Mon`
`17:23.45`	`Strom_C`	`i was asking you why they need this weirdo automatic conference feature in the first place`
`17:24.01`	`De_Mon`	`thats the way they do business`
`17:24.19`	`chiang_sg`	`party chat line ?`
`17:24.28`	`Strom_C`	`which is a roundabout way of saying that you really have no idea, right?`
`17:24.56`	`De_Mon`	`they do support, and regularly have multiple people in the support call, often times 3+`
`17:25.50`	`De_Mon`	`sometimes it turns into a meeting of 5-9 people. But the ptb want the conference ended if either the caller of called person leave the conference`
`17:25.58`	`De_Mon`	`*or`
`17:26.31`	`Strom_C`	`so how about doing this instead`
`17:26.42`	`Strom_C`	`calling party dials internal extension number from the IVR menu`
`17:26.56`	`Strom_C`	`calling party generates a .call file and dumps into a meetme as a marked user`
`17:27.15`	`Strom_C`	`called party gets called when the .call file is processed, answers, and also gets dumped into a meetme as a marked user`
`17:27.42`	`Strom_C`	`other parties dial into a different extension so as not to be marked`
`17:28.29`	`markit`	`hi, is theoretically possible run asterisk under a VM, like with KVM? or there are performance/responsiveness issues too bad?`
`17:29.44`	`*** join/#asterisk Skymarshal (n=Skymarsc@pD9E131FD.dip0.t-ipconnect.de)`
`17:29.45`	*** join/#asterisk ez` (n=Ez@c66.203.210-59.clta.globetrotter.net)
`17:30.33`	`Skymarshal`	`I build zaptel-1.4 and asterisk-1.4 but after modprobe ztdummy and starting asterisk I can not use MeetMe(). Same way works with version 1.2. Any ideas?`
`17:31.08`	`*** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net)`
`17:31.45`	`*** join/#asterisk potential (n=anthony@175.21.188.72.cfl.res.rr.com)`
`17:32.19`	`potential`	`Hello, anyone awake on this beautiful thursday afternoon? :)`
`17:33.36`	`*** join/#asterisk Zodiacal (n=hehehe@bdsl.66.14.242.199.gte.net)`
`17:33.56`	`Qwell[]`	`potential: ...`
`17:33.58`	`Qwell[]`	`~unlimited`
`17:34.09`	`jbot`	`rumour has it, unlimited is <Nugget> unlimited voip == punch the monkey to win a free ipod`
`17:34.11`	`Qwell[]`	`stupid slow bot, I swear`
`17:35.03`	ez`	`my asterisk is behing a debian gateway box ; do i need to foward a port to enable extenal iax client to reach my asterisk ; ??`
`17:35.27`	`*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)`
`17:35.39`	ez`	`i beleiev yes`
`17:35.50`	`De_Mon`	`Strom_C how does that end the conference if the person who initiated the conference leaves?`
`17:35.53`	`*** join/#asterisk mwbgrob (n=mgrob@82.79.21.128)`
`17:36.30`	`Strom_C`	`De_Mon: it was a general suggestion for you to tweak to your needs`
`17:36.30`	`*** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com)`
`17:36.39`	`Strom_C`	`but it works around the caller ID problems you were having`
`17:36.42`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`17:36.43`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`17:37.00`	`potential`	`~wikis`
`17:37.02`	`jbot`	`wikis is, like, http://www.voip-info.org`
`17:37.08`	`Qwell[]`	`oh, sure, be fast for him`
`17:39.28`	`Chris-NB`	`hi`
`17:39.36`	`Chris-NB`	`anyone knows what ISDN Timer t200 is?`
`17:42.14`	`markit`	`Chris-NB: no, but I'm happy to find one that has isdn and asterisk :) do you use mISDN?`
`17:43.30`	`potential`	`where do I find a list of providors for free incoming and outgoing?`
`17:44.17`	`cygar`	`hello`
`17:45.27`	`Strom_C`	`today's dilbert wins big: http://www.dilbert.com/comics/dilbert/archive/images/dilbert20071832660125.gif`
`17:46.19`	`*** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net)`
`17:46.33`	`*** join/#asterisk RoyK (n=roy@sbs.hardrockcafe.no)`
`17:46.52`	`cygar`	`I am sending dtmf through rfc2833 in sip. Is there an option [with verbose and sip debug peer XX] where i can see the digits [dtmf] pressed by some extension? I'm having problems with dtmf "1" and not sure if it's getting to the * box or is having a problem when sending it to the e1 card [r2]`
`17:47.40`	`cygar`	`i remember i used to see the dtmf in the logs when using verbose or sip debug...`
`17:48.09`	`JunK-Y`	`cygar: rtp debug?`
`17:48.12`	`*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)`
`17:48.26`	`*** join/#asterisk QbY (n=Kelvin@66.236.241.67.ptr.us.xo.net)`
`17:48.58`	`cygar`	`JunK-Y: when sending through rfc2833 the dtmf are sent in rtp as "inband" ?`
`17:49.31`	`potential`	`vtnoc.net good?`
`17:49.45`	`cygar`	`JunK-Y: i know, i should read the rfc. I think i used to see it cause i was using dtmf in INFO [ sip ]`
`17:51.06`	`*** join/#asterisk nakee (n=nakee@grok.cs.huji.ac.il)`
`17:51.07`	`*** join/#asterisk RoyK (n=roy@sbs.hardrockcafe.no)`
`17:51.31`	`brad_mssw`	`upgrading from asterisk 1.2.12 -> 1.2.14 jacked up my music on hold, it's loud as hell ... granted mpg123 was also upgraded for 0.59r (think that was the version), to 0.61`
`17:52.24`	`*** join/#asterisk nvicf (n=nvicf@201.250.161.32)`
`17:53.02`	`wunderkin`	`you should only use mpg123 0.59r, or rather i recommend native, unless you have a reason to use mpg123 instead`
`17:57.13`	`*** join/#asterisk defswork (n=andy@mailgate2.3gcomms.co.uk)`
`17:59.05`	`nvicf`	`can I use asterisk to answer my door?like a doorman?weird question, I need to do it as if it were just an extra telephone line?`
`18:00.02`	`brad_mssw`	`wunderkin: native doesn't decode mp3, does it ?`
`18:00.13`	`brad_mssw`	`wunderkin: you'd have to convert those to ulaw or whatnot first, right?`
`18:00.21`	`wunderkin`	`no you convert your files to whatever codec you want, do you always use the same one?`
`18:00.36`	`brad_mssw`	`wunderkin: well, we have some g729 and some ulaw ... that's it though`
`18:00.52`	`*** join/#asterisk shinux__ (n=shinux@196.220.26.237)`
`18:06.45`	`*** join/#asterisk faberk64 (n=faberk@213.199.15.249)`
`18:07.30`	`*** join/#asterisk XIN01OZ (n=KNOCK@c-68-63-34-189.hsd1.al.comcast.net)`
`18:07.39`	`elriah`	`Ok, I'm dialing an IAX2 host. It connects to the default context just fine. How can I tell the call to go to another context? I tried to put context=whatever in the iax2.conf user definition but it always wants to go to s@default.`
`18:08.02`	`Bobthehunter`	`http://pastebin.ca/328294`
`18:08.24`	`*** part/#asterisk bricecubed (n=nesta@pool-72-84-202-204.rcmdva.east.verizon.net)`
`18:08.50`	`elriah`	`Rather, how do I force a context on an inbound IAX trunk?`
`18:09.31`	`*** join/#asterisk trevarthan (n=trevarth@c-71-59-48-26.hsd1.ga.comcast.net)`
`18:09.46`	`Dr-Linux`	`what this warning means: chan_iax2.c:691 jb_warning_output: Resyncing the jb. last_delay -5, this delay 1537, threshold 1062, new offset -6556`
`18:09.46`	`Dr-Linux`	`Jan 25 10:00:31 WARNING[25129]: chan_iax2.c:7971 network_thread: chan_iax2: ast_sched_runq ran 49 scheduled tasks all at once`
`18:10.04`	`trevarthan`	`hello. what sort of CPU/RAM requirements are recommended for 48 simultaneous calls over a dual T1 card?`
`18:10.30`	`Qwell[]`	`trevarthan: shouldn't need a whole heck of a lot of power, especially if you do hardware echo can`
`18:10.51`	`trevarthan`	`does digium hardware do that?`
`18:11.10`	`Qwell[]`	`trevarthan: If you buy the module, yes`
`18:11.12`	`bkw_`	`the correct question is can it do it very well ;)`
`18:11.19`	`Qwell[]`	`I think the dual port has it`
`18:11.57`	`elriah`	`Hey Qwell, how do you force a context on an inbound iax peer?`
`18:12.05`	`trevarthan`	`do you think a 2.8ghz would be enough?`
`18:12.06`	`*** join/#asterisk [hC] (n=hardcore@66.119.167.162)`
`18:12.10`	`trevarthan`	`P4.`
`18:12.14`	`Qwell[]`	`trevarthan: should be`
`18:12.22`	`[hC]`	`morning people`
`18:12.27`	`[hC]`	`Qwell[]: want a 7970 yet? :)`
`18:12.36`	`Qwell[]`	`[hC]: not yet...`
`18:12.51`	`Qwell[]`	`and, technically, there is one sitting...somewhere in this office`
`18:13.08`	`Qwell[]`	`I assume it's got skinny firmware - I haven't actually checked`
`18:13.13`	`[hC]`	`well... if you want mine... just let me know`
`18:13.19`	`[hC]`	`I'll unplug the one off my desk and send it along`
`18:13.21`	`De_Mon`	`Strom_C I worked around the callerID issue another way, my orignal question was how to kick everyone of of a conference when one OR another user left.`
`18:13.21`	`Qwell[]`	`"want" ;)`
`18:13.27`	`Dr-Linux`	`Qwell[]: any clue about my question?`
`18:13.47`	`[hC]`	`I'm using skinny ..... 7.0.1?`
`18:13.53`	`Qwell[]`	`sounds about right`
`18:13.55`	`[hC]`	`TERM70.7-0-1-0s`
`18:14.02`	`Qwell[]`	`I think that's what I was using last time I messed with a 7970 at WF`
`18:14.13`	`Qwell[]`	`maybe 7.0.3 or something`
`18:14.29`	`[hC]`	`Ive kept cisco firmware behind on all my devices, Skinny and SIP... after my last experience`
`18:14.36`	`Qwell[]`	`good thinking ;)`
`18:14.53`	`[hC]`	`I upgraded the SIP firmware and it introduced a bug where callers would lose audio for approximately 5 seconds when dhcp renewed, or something stupid`
`18:14.57`	`[hC]`	`The ones i have work. They stay.`
`18:14.58`	`[hC]`	`:)`
`18:14.59`	`Qwell[]`	`nice`
`18:15.15`	`De_Mon`	`where should I send channels in the h extension when I want to stop processing them`
`18:15.28`	`trevarthan`	`does the Digium Wildcard TE205P have hardware echo cancel?`
`18:15.33`	`De_Mon`	`instead of h,n(hangup),Hangup()`
`18:15.38`	`Qwell[]`	`trevarthan: it can - if you get the module`
`18:15.49`	`Qwell[]`	`then it technically becomes a TE207P or something`
`18:16.01`	`Qwell[]`	`however`
`18:16.04`	`trevarthan`	`Is that the Digium Wildcard TE207P with the module?`
`18:16.04`	`De_Mon`	`maybe a NoOp()? that seems verbose.`
`18:16.07`	`Qwell[]`	`trevarthan: not everybody needs echo can`
`18:16.09`	`trevarthan`	`right`
`18:16.15`	`trevarthan`	`do I need it?`
`18:16.16`	`Qwell[]`	`trevarthan: so, I'd try without it, and see how it goes`
`18:16.28`	`bkw_`	`Qwell[], If Asterisk and Digium are so Open Source why do they not release the Lumenvox connector source code?`
`18:16.43`	`Qwell[]`	`bkw_: Now you're just trolling.`
`18:16.44`	`bkw_`	`Qwell[], they have the power to grant the exception for the code...`
`18:16.47`	`bkw_`	`no i'm not`
`18:16.51`	`bkw_`	`honestly I want to know`
`18:17.09`	`Qwell[]`	`like I said last night - I don't know the agreement we have with Lumenvox. If you'd like to know, you'll have to call sales or something`
`18:17.24`	`elriah`	`Anyone? How do I construct a dial() command to include the context? I've tried Dial(IAX2/somehost/s@somecontext) but it defaults to s@default everytime ...`
`18:17.37`	`Qwell[]`	`whatever they tell you - feel free to make it public, but the fact of the matter is, I don't know the answer, so I don't even want to entertain a guess`
`18:18.06`	`bkw_`	`Qwell[], i'll email sales then ;)`
`18:18.58`	`*** join/#asterisk justinc- (n=samblack@host-64-179-18-177.spr.choiceone.net)`
`18:19.05`	`trevarthan`	`Qwell[]: does asterisk benefit from dual processors?`
`18:19.15`	`Qwell[]`	`trevarthan: yes, it's heavily threaded`
`18:19.19`	`De_Mon`	`elriah dialing a new host? the host decides what context the call goes into`
`18:19.33`	`trevarthan`	`in other words, would it be smarter to just buy dual CPUs and the cheaper card if that's cheaper?`
`18:19.38`	`bkw_`	`Qwell[], you forgot to mention that its scared of threads`
`18:19.44`	`*** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir)`
`18:19.49`	`bkw_`	`it uses them but doesn't take full advantage where it COULD`
`18:19.52`	`De_Mon`	`elriah tell your host that calls from you belong in a specific context and go from there.`
`18:19.55`	`Qwell[]`	`trevarthan: possibly - it's just something you'll have to test yourself, and come up with a proper solution`
`18:19.56`	`bkw_`	`more threads the better`
`18:19.59`	`Qwell[]`	`trevarthan: it would, however, work`
`18:20.06`	`elriah`	`Right, but where do I set that (on the host side) I tried context=whatever in the iax.conf user definition. Always wants to default to s@default. (Thanks for the help, btw)`
`18:20.32`	`elriah`	`De_Mon: So on my peer iax.conf entry, specify context=whatever there?`
`18:21.05`	`De_Mon`	`elriah do you have a guest context where context=default?`
`18:21.15`	`elriah`	`Nope. Removed it.`
`18:21.47`	`elriah`	`I actually don't have a default context at all...`
`18:21.47`	`De_Mon`	`elriah did you reload iax after making changes to iax.conf?`
`18:21.51`	`elriah`	`Yep.`
`18:21.59`	`*** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br)`
`18:22.01`	`De_Mon`	`elriah something is trying to send your caller there...`
`18:22.11`	`elriah`	`I get this: Rejected connect attempt from xx.xx.xx.xx, request 's@default' does not exist`
`18:22.12`	`De_Mon`	`having it doesn't change where it tries to go :)`
`18:22.22`	`elriah`	`hrm...`
`18:22.34`	`elriah`	`Anywhere besides my extensions.conf or iax.conf that would be?`
`18:22.55`	`De_Mon`	`try not using the @ sign, I don't think IAX knows about contexts in dial like that`
`18:23.29`	`rene-`	`hello guys anyone can please give me a ring at my mexican number, i think my internationally connectivity is down, just a quick hi 52 9982874123`
`18:23.30`	`elriah`	`I'm not specifying anything in that example, no context at all, and s@default is what asterisk is telling me is invalid.`
`18:24.07`	`De_Mon`	`what is the dial command you are using when that error occurs`
`18:24.11`	`elriah`	`If I put a default context in there and then just do a goto, it works fine.`
`18:24.56`	`elriah`	`dial(IAX2/myiaxconfuser:mysecret@my.fqdn)`
`18:25.08`	`*** join/#asterisk J4k3- (i=jsuter@dhcp-12-197-128-58.intrastar.net)`
`18:25.31`	`elriah`	`I'd like to be able to do something like: dial(IAX2/myiaxconfuser:mysecret@my.fqdn/s@somecontext)`
`18:26.26`	`De_Mon`	`hrm, it should be doing s@whateverdefaultcontextissetto`
`18:26.37`	`elriah`	`Right. That's what I thought.`
`18:27.02`	`Dr-Linux`	`Qwell[]: i have simliar problem >> http://threebit.net/mail-archive/asterisk-users/msg00013.html`
`18:27.04`	`elriah`	`If I specify any valid context, it just returns the error.`
`18:27.14`	`elriah`	`If I don't specify a context and my default context exists, works grat.`
`18:27.17`	`elriah`	`great.`
`18:27.24`	`Dr-Linux`	`i'd appreciate any help`
`18:27.25`	`De_Mon`	`when you dial a resource there is no context, just the extension`
`18:28.00`	`De_Mon`	`default context is set in iax.conf globaly or specific for that user/peer (whatever)`
`18:28.01`	`elriah`	`.. in one set of documentation, which is fine by me, then I should be able to control it with my iax.conf peer/user definition with context=whatever, right?`
`18:28.17`	`De_Mon`	`yeah`
`18:28.25`	`justinc-`	`the context in aix and sip .confs are for outbound dialing...`
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`18:28.45`	`De_Mon`	`buwah?`
`18:28.57`	`De_Mon`	`dundun duuuuun`
`18:28.57`	`elriah`	`So if my peer's iax.conf has a context=whatever, it's going to try to find the context 'whatever' on the host it's calling?`
`18:29.26`	`elriah`	`split..@$`
`18:29.26`	`De_Mon`	`:/`
`18:30.16`	`sevard`	`really?`
`18:30.16`	`Qwell[]`	`That was hardly a split`
`18:30.17`	`sevard`	`i thought a bunch of people just decided to quit`
`18:30.18`	`sevard`	`silly me.`
`18:30.18`	`justinc-`	`no it'll try to dial from context=whatever on the machine it's registered to`
`18:30.19`	`Qwell[]`	`that was like...nobody`
`18:30.19`	`justinc-`	`what just happened?`
`18:30.19`	`De_Mon`	`context= is the context in extensions.conf where the received call goes on that machine`
`18:30.31`	`justinc-`	`no I think it's the other way around`
`18:30.37`	`elriah`	`So again, in host A (The "Client") calling via iax2 host B (The "Server"), host B's iax.conf entry should have context=whatever and anyone calling to that iax.conf peer/user definition should fall into that context?`
`18:30.38`	`justinc-`	`unless you're talking about zap channel config`
`18:30.57`	`De_Mon`	`you can set an inbound context for peers`
`18:31.04`	`elriah`	`Right. Where?`
`18:31.09`	`justinc-`	`oh .. that's right good point`
`18:31.09`	`elriah`	`Who's on first?`
`18:31.19`	`De_Mon`	`I dont know is on first!`
`18:31.23`	`De_Mon`	`who's at home`
`18:31.24`	`elriah`	`THIRD BASE!`
`18:31.28`	`elriah`	`bahahaha`
`18:31.55`	`De_Mon`	`http://www.voip-info.org/wiki/view/Asterisk+config+iax.conf`
`18:32.10`	`*** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg)`
`18:32.12`	`justinc-`	`I'm used to sip as opposed to iax`
`18:32.17`	`skirmisha`	`file hello again`
`18:32.48`	`skirmisha`	`for asterisk ver 1.4 do i need to download trunk in order to have this rtp payload manipulation`
`18:32.57`	`skirmisha`	`or in branches is also supported`
`18:33.02`	`elriah`	`Got it. It works now. Thanks for the help.`
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`18:34.13`	`justinc-`	`I have a bit of a complex enigma dealing with the new dynamic agents if anyone's willing to help me take a shot at it:`
`18:34.58`	`justinc-`	`I've built a proxy for asterisk (amsuite.sourgeforge.net) for call center stats and such. I'm trying to dynamically map a dynamic agent on a Local/ channel to the phone's real channel and can't think of a good way to do it`
`18:35.17`	`justinc-`	`any ideas?`
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`18:38.24`	`skirmisha`	`justinc- what's the idea`
`18:38.39`	`justinc-`	`I'm sorry? :)`
`18:38.50`	`justinc-`	`what am I trying to do you mean?`
`18:38.57`	`skirmisha`	`yes`
`18:39.02`	`justinc-`	`oh ok`
`18:39.04`	`justinc-`	`well...`
`18:39.14`	`justinc-`	`asterisk did away with "agents" as we knew them`
`18:39.45`	`skirmisha`	`yes`
`18:39.58`	`justinc-`	`so now dynamic agents run the show. My proxy tracks agents making outbound calls, but it doesn't know how to tell that SIP/300 is logged in as Local/300@agent-contact`
`18:40.01`	`justinc-`	`for example :)`
`18:40.09`	`Qwell[]`	`there is a doc in 1.4 that explains how you should do dynamic queue members`
`18:40.11`	`justinc-`	`I'm trying to figure out a way to map them ..`
`18:40.23`	`justinc-`	`no no I know how to make dynamic members work...`
`18:40.42`	`justinc-`	`but digium destroyed the agent encapsulation that makes working with agents "easy"`
`18:40.46`	`anonymouz666`	`very very strange.... the serveremail does not set correctly the hostname on mail from when sending voicemail`
`18:40.55`	`Qwell[]`	`justinc-: the doc in 1.4 explains it`
`18:40.55`	`anonymouz666`	`why?`
`18:41.00`	`elriah`	`Do agi scripts have to be called from the agi-bin directory? Can they be called from any system path?`
`18:41.07`	`justinc-`	`I have read the docs`
`18:41.14`	`Qwell[]`	`elriah: should be able to, if you give it a full path`
`18:41.14`	`justinc-`	`maybe I'm not explaining this correctly ...`
`18:41.23`	`elriah`	`k, thanks, Qwell.`
`18:41.39`	`justinc-`	`as per 1.4 docs you log an agent in say as Local/${EXTEN}@context`
`18:41.41`	`justinc-`	`right?`
`18:41.45`	`Corydon-w`	`justinc-: the problem with the old AgentCallbackLogin was that it was fraught with so many race conditions, it was better to get rid of it`
`18:41.51`	`Qwell[]`	`justinc-: yes`
`18:42.02`	`justinc-`	`yes I know I've dealt with the issues myself :)`
`18:42.05`	`justinc-`	`anyway ...`
`18:42.14`	`xpot`	`anyone know how to output a number VAL=60.000000 when performing MATH func to just a whole value ex: VAL=60 ?`
`18:42.17`	`justinc-`	`if an agent makes a call directly from their phone .. say SIP/300`
`18:42.43`	`justinc-`	`there's no way to know that the actual phone belongs to Local/300@context`
`18:42.46`	`Corydon-w`	`xpot: int type`
`18:42.47`	`justinc-`	`get what I mean?`
`18:43.32`	`xpot`	`Corydon: how do I specify in the MATH func?`
`18:43.54`	`Corydon-w`	`justinc-: no intrinsic way, but you always have to make assumptions in your dialplan, because you wrote it`
`18:44.02`	`justinc-`	`that's the thing`
`18:44.18`	`xpot`	`here is what I have: exten => s,n,Set(MINREM=${MATH(${RSEC} / 60)})`
`18:44.28`	`justinc-`	`this isn't for asterisk .. it's for the proxy software.. and I can't make assumptions about a dialplan because other people are using it`
`18:44.36`	`Corydon-w`	`xpot: ", int" at the end`
`18:44.42`	`justinc-`	`so I'm just trying to figure out some way to map them`
`18:44.45`	`xpot`	`ok, thank you`
`18:45.10`	`Qwell[]`	`justinc-: the same way you would have with agents`
`18:45.20`	`Corydon-w`	`xpot: type: show function MATH`
`18:45.21`	`justinc-`	`I was hoping someone might have an idea that would avoid making people write another conf file that maps Local channels to actual phone channels ... which would be annoying to the people using the software`
`18:45.26`	`Corydon-w`	`xpot: it's right in the example`
`18:45.48`	`justinc-`	`Qwell[]: What do you mean?`
`18:45.51`	`Qwell[]`	`an "agent" didn't make outbound calls previous`
`18:45.53`	`Qwell[]`	`previously`
`18:45.56`	`xpot`	`sorry, I must have read it too fast`
`18:45.58`	`justinc-`	`correct a phone does`
`18:46.07`	`Qwell[]`	`so, you'd do the mapping the same way you did before`
`18:46.09`	`justinc-`	`however it was trivial to map an agent to a phone channel`
`18:46.10`	`Qwell[]`	`however that was`
`18:46.14`	`justinc-`	`now it isn't`
`18:46.45`	`Qwell[]`	`How did the map them before?`
`18:46.45`	`Qwell[]`	`How did you know that SIP/200 was Agent/Bob?`
`18:46.45`	`skirmisha`	`anonymouz666 check how u have created asterisk account in passwd`
`18:46.45`	`justinc-`	`agent channels are forced to be numbers`
`18:46.45`	`justinc-`	`so it was easier`
`18:46.50`	`Qwell[]`	`okay then`
`18:46.52`	`Qwell[]`	`How did you know that SIP/200 was Agent/400?`
`18:46.53`	`Corydon-w`	`justinc-: there's no such assumption in the code`
`18:46.58`	`xpot`	`ok, that is different.. I pulled the info from voip-info didn't see the type-of-result there`
`18:47.11`	`xpot`	`I will use the show function instead`
`18:47.15`	`justinc-`	`oh because Agents have "Locations" when logged in`
`18:47.16`	`xpot`	`thanks`
`18:47.25`	`justinc-`	`I didn't need to do any hacking or assumptions`
`18:47.37`	`justinc-`	`the actual channel of the PHONE was an argument to agentcallbacklogin`
`18:47.42`	`justinc-`	`no assumptions needed :D`
`18:47.44`	`Corydon-w`	`justinc-: actually, you DID make an assumption`
`18:47.49`	`justinc-`	`what's that?`
`18:48.02`	`Corydon-w`	`justinc-: you assumed that the agent number matched the channel number`
`18:48.05`	`justinc-`	`no`
`18:48.06`	`justinc-`	`again`
`18:48.14`	`justinc-`	`the actual phone channel was an argument to agentcallbacklogin`
`18:48.18`	`justinc-`	`no assumptions made at all`
`18:48.36`	`justinc-`	`however with dynamic agents this doesn't work at all`
`18:48.43`	`Corydon-w`	`justinc-: sure it does`
`18:48.47`	`justinc-`	`because it's completely free form with no agent->channel linkage`
`18:49.01`	`justinc-`	`Corydon-w: explain...`
`18:49.03`	`Corydon-w`	`justinc-: which you define`
`18:49.18`	`Corydon-w`	`you define the linkage in the dialplan`
`18:49.25`	`*** join/#asterisk Assid (i=assid@59.183.31.171)`
`18:49.27`	`justinc-`	`uh huh...`
`18:49.28`	`Assid`	`heya`
`18:49.38`	`justinc-`	`but that's fine`
`18:49.55`	`justinc-`	`in the old way the dialplan would log an agent in using their phone channel`
`18:49.58`	`justinc-`	`so it all worked out`
`18:49.58`	`Assid`	`anyone using freecall/sipdiscount or those mirror companies?`
`18:50.02`	`Corydon-w`	`VMAuthenticate establishes which Local channel to use`
`18:50.24`	`Corydon-w`	`(in the example provided)`
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`18:51.29`	`J4k3`	`sipdiscount = * Max 300 minutes per week of free calls, measured over the last 7 days and per unique IP address.`
`18:51.35`	`Assid`	`yeah`
`18:51.59`	`justinc-`	`hmm I guess there's just no way around it ... the channels have to be mapped in a config file .. /sigh that kinda sucks :(`
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`18:52.58`	`Corydon-w`	`justinc-: I don't see why they have to be mapped anywhere`
`18:53.20`	`justinc-`	`well the whole point is to keep outgoing statistics for agents`
`18:53.44`	`Dr-Linux`	`anybody know about this warning: WARNING[19418]: chan_iax2.c:691 jb_warning_output: Resyncing the jb. last_delay 5, this delay 15876, threshold 1068, new offset -15876`
`18:54.01`	`justinc-`	`if an agent is logged in under a channel (like a Local channel) that isn't specifically their phone channel .. there's no way of keeping stats on that agen'ts outgoing call`
`18:54.13`	`justinc-`	`so they have to be mapped...`
`18:54.22`	`Corydon-w`	`Ah, you're worried about statistics`
`18:54.25`	`Dr-Linux`	`is it bandwidth issue?`
`18:54.26`	`justinc-`	`yep :)`
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`18:55.04`	`Corydon-w`	`Well, then you track statistics via CDR`
`18:55.18`	`justinc-`	`nope .. real time via events ...`
`18:55.25`	`justinc-`	`this is 3rd party stats`
`18:55.32`	`justinc-`	`er 3rd party software I mean`
`18:55.38`	`sevard`	`I wish there was a way to run voip over 4Kb/s :\|`
`18:55.44`	`Assid`	`sure you can`
`18:55.49`	`Assid`	`use gsm`
`18:55.50`	`sevard`	`KB*`
`18:55.53`	`Strom_C`	`sevard: it's called lpc10`
`18:56.03`	`sevard`	`ilpc will run over 4KB?`
`18:56.12`	`Strom_C`	`not ilbc`
`18:56.14`	`Strom_C`	`lpc10`
`18:56.20`	`Strom_C`	`it's 2.4kbps for the codec`
`18:56.20`	`Corydon-w`	`Mr Roboto`
`18:56.32`	`Strom_C`	`haha`
`18:56.33`	`sevard`	`It's got to sound like ass`
`18:56.35`	`Strom_C`	`well, when you're desperate...`
`18:57.01`	`Dr-Linux`	`Strom_C: any clue for my question?`
`18:57.30`	`sevard`	`since I don't have money for PSTN termination I was thinking about hooking up an * box and then hooking up my parents and my girlfriend, but my girlfriend wanted her mom to be on the system... but her mom has shitty dialup`
`18:57.35`	`Strom_C`	`Dr-Linux: yeah, you've got severe jitter`
`18:57.57`	`Dr-Linux`	`Strom_C: is that a bandwidth issue?`
`18:58.04`	`Jingles`	`I've been drinking rockstars all morning. I've got severe jitter too.`
`18:58.05`	`justinc-`	`Cordyon-w: what do you think about this idea? When the proxy gets an addQueueMember event, it can call the Location (channel) it gets and track for a Dial event to record the channel that gets dialed and map it that way ...`
`18:58.17`	`Dr-Linux`	`Strom_C: or i need some settings for iax?`
`18:58.33`	`sevard`	`Strom_C: Have you ever tried lpc10 over a dialup connection?`
`18:58.47`	`Strom_C`	`no`
`18:59.08`	`sevard`	`I'm going to have to give it a shot.`
`18:59.14`	`Strom_C`	`it's been so long since i've used a dialup connection....`
`18:59.21`	`Strom_C`	`sevard: let me know how it works out`
`18:59.30`	`Strom_C`	`how shitty is the dialup?`
`18:59.31`	`sevard`	`will do, i just need to find a colo for my * box`
`18:59.46`	`Dr-Linux`	`Strom_C: should i change my codecs or what?`
`18:59.48`	`sevard`	`it's shitty enough that it'll run at about 3-4 KB/s`
`18:59.53`	`Strom_C`	`Dr-Linux: I don't know`
`18:59.59`	`Dr-Linux`	`ok`
`19:00.05`	`Strom_C`	`sevard: oh, kiloBYTES per second`
`19:00.16`	`sevard`	`yeah, bytes, not bits`
`19:00.40`	`sevard`	`i'm taking anywhere between 29-35,000 bits`
`19:00.48`	`Strom_C`	`so it's roughly a 33.6kbps modem`
`19:00.52`	`Strom_C`	`yeah, ok`
`19:00.58`	`sevard`	`well, it's a 56k, but the line quality is such`
`19:01.13`	`Strom_C`	`56k modems are always asymmetric though`
`19:01.40`	`sevard`	`hmm`
`19:01.52`	`Strom_C`	`even with 28.8kbps upstream though, you should be fine with iax2 and gsm or ilbc`
`19:02.17`	`J4k3`	`for interactive activities v.34 beats v.90`
`19:02.26`	`J4k3`	`v.92 on a perfect line beats both, but perfect lines don't exist in the real world`
`19:02.27`	`sevard`	`really, i thought the overhead would kill a 56k connection`
`19:02.58`	`Strom_C`	`nope....13kbps for gsm audio + 9.6kbps iax2 overhead`
`19:03.13`	`sevard`	`sweetness.`
`19:03.32`	`J4k3`	`Strom_C: well the bigger issue is that modems are asynchronous and they do have a tx/rx switchover time`
`19:03.36`	`sevard`	`i'll give it a shot, do you know if xlite supports lpc-10 if I run out of options?`
`19:03.55`	`J4k3`	`50 pps is quite a challenge for a modem`
`19:04.35`	`*** join/#asterisk BB\|AtWork (n=karl@38.99.18.98)`
`19:05.09`	`BB\|AtWork`	`include => a_context_here would run the [a_context_here] script that follows it right?`
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`19:06.09`	`naftali5`	`BB\|AtWork, not really, it would jutst make the extensions within that context available to the context it was included in`
`19:06.17`	`sevard`	`J4k3: so you're saying it wouldn't work out simply because of the delay between tx/rx`
`19:06.50`	`J4k3`	`sevard: no, I'm just saying its a bit of a challenge ;)`
`19:06.57`	`J4k3`	`it can work, theres no doubt`
`19:07.11`	`J4k3`	`its just going to require tweaking to get it happy`
`19:07.19`	`sevard`	`LPC10 Total: 54 bits per frame, 2400 bps`
`19:07.21`	`BB\|AtWork`	`naftali5, hrm well in this case its a handler for just call forwarding (all extensions *73)`
`19:07.25`	`*** join/#asterisk dasenjo (n=dasenjo@190.5.196.105)`
`19:07.37`	`*** join/#asterisk Legend (n=Legend@office.bgcfreedom.com)`
`19:07.55`	`Legend`	`has anyone used these net2phone voicedirector things?`
`19:08.00`	`Legend`	`seem to be built on asterisk`
`19:08.20`	`*** join/#asterisk Manfish (n=themanfi@82-68-173-121.dsl.in-addr.zen.co.uk)`
`19:08.23`	`naftali5`	`BB\|AtWork, so if you define the exten=>73 in that context, then dialing 73 in the original context will run that`
`19:09.12`	`BB\|AtWork`	`hrm. thats what i thought. but its not working`
`19:09.15`	`BB\|AtWork`	`must be something else`
`19:09.37`	`Manfish`	`anyone here able to shed any light on a moh problem where the track restarts after each announcement in a queue instad of pausing and picking up where it left off`
`19:09.51`	`sevard`	`who wants to host my pbx? :\|`
`19:14.02`	`*** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch)`
`19:17.26`	`*** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com)`
`19:22.11`	`*** join/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net)`
`19:22.59`	`*** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)`
`19:25.15`	`*** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no)`
`19:25.36`	`tzafrir_laptop`	`hi, anybody here connecting to BRI inFrench? preferably with ZapBRI? I'm trying to debug a connection`
`19:25.55`	`tzafrir_laptop`	`I'm not exactly sure about the settings`
`19:26.21`	`rikstah`	`tzafrir, im in UK, is it euroisdn?`
`19:26.46`	`tzafrir_laptop`	`yeah, euroisdn`
`19:26.56`	`rikstah`	`ok i guess i can help, whats the prob`
`19:26.57`	`*** join/#asterisk bkruse (i=bkruse@nat/digium/x-ac5b61d32ba41d85)`
`19:27.12`	`bkruse`	`anyone here help me with a php problem im having with sessions? just conceptual problem :[`
`19:27.36`	`*** join/#asterisk HushPe (n=HushPe@mail.kamar.co.nz)`
`19:28.04`	`tzafrir_laptop`	`in zaptel.conf I have span=1,1,1,ccs,ami`
`19:28.34`	`*** join/#asterisk hassler (n=hassler@r-corp.hcst.com)`
`19:28.36`	`rikstah`	`here i have ccs,hdb3,crc4`
`19:28.45`	`rikstah`	`thats for isdn30/euroisdn`
`19:28.54`	`*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)`
`19:29.00`	`tzafrir_laptop`	`that's PRI, isn't it?`
`19:29.17`	`tzafrir_laptop`	`anyway, I'll give it a shot`
`19:29.21`	`rikstah`	`E1`
`19:29.23`	`hassler`	`hi folks, new trixbox installation. sip phones can hear MOH, but not recordings. Quick, what did I miss?`
`19:29.26`	`rikstah`	`europe`
`19:29.58`	`tzafrir_laptop`	`Though if I had such wrong settings I would have had alarms on the span, I believe (yellow or blue?)`
`19:30.14`	`rikstah`	`tzafrir_laptop, i woulda have thought so too...but who knows ;)`
`19:30.41`	`rikstah`	`tzafrir_laptop, i'm just telling you what i have here for UK (which i believe is the same)`
`19:31.33`	`tzafrir_laptop`	`I'm also not sure BRI and PRI are the same even within the same telco. Not to mention the UK tends to be "different"...`
`19:31.44`	`sevard`	`http://voipgear.blogspot.com/2007/01/new-phones-from-aastra-on-way.html`
`19:31.46`	`sevard`	`Oooooooooooooooo pretty`
`19:32.11`	`De_Mon`	`how can I end a conference when either marked user leaves instead of both marked users leave`
`19:32.37`	`CPSK`	`Hi, looking for help with Tenovis Integral E1 and TE110P.. msg me, thanx`
`19:32.38`	`sevard`	`De_Mon: killall -9 asterisk`
`19:32.50`	`sevard`	`CPSK: Dude, shut the _fuck_ up.`
`19:33.04`	`rudholm`	`sevard: it's automated`
`19:33.11`	`rudholm`	`no point responding`
`19:33.14`	`sevard`	`De_Mon: I hope you knew that was a joke.`
`19:33.26`	`sevard`	`rudholm: I thought we banned him, and he changed the timer to a /say instead of a /me`
`19:33.31`	`sevard`	`that means he has to at least be around`
`19:33.33`	`Manfish`	`nice new phones, screens are cetting bigger than my laptop :)`
`19:33.35`	`tzafrir_laptop`	`BTW: any point at all to play with the LBO parameter? Or is it only for small analog changes?`
`19:33.44`	`HushPe`	`is it possible to set a context for my zap channels? so some lines are for some depts, and others for another?`
`19:33.55`	`rikstah`	`tzafrir_laptop, to be honest, i have no idea :)`
`19:34.04`	`HushPe`	`or do i pick that up within a context and then change the context in the extensions file?`
`19:34.35`	`mercestes`	`he also fixed his typeo`
`19:34.44`	`tzafrir_laptop`	`HushPe, surprisingly, this is done with the setting "context=custontext"`
`19:35.34`	`HushPe`	`i know how to do it ;) it's just where to do it (like correct place)`
`19:35.34`	`HushPe`	`i don't usually pull out my idiot hat until later on in the day (it's 8:30am local time)`
`19:35.50`	`sevard`	`1:30 P.M. here`
`19:35.58`	`sevard`	`my idiot hat is well under way`
`19:36.15`	`J4k3`	`I keep my idiot bottled up, and only pour it out after hours.`
`19:36.18`	`HushPe`	`sevard: ;)`
`19:36.42`	`mercestes`	`HushPe: It's somethign like bchan=1-5 group=1 context=catsex and bchan=6-10 group=2 context=wtf I think.`
`19:36.43`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`19:36.44`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`19:37.02`	`HushPe`	`mercestes: cool :)`
`19:37.17`	`mercestes`	`be sure you use the context catsex or the whole thing won't work`
`19:37.20`	`[hC]`	`When doing an attended transfer in asterisk (I have # set up to transfer attended) - When the transfer is complete, it beeps to the transferrER that the transfer is complete. Is there a way to indicate to the person who has just had a call transferred to them that they are now connected with the caller?`
`19:37.50`	`mercestes`	`[hC]: The real answer is yes. The practical answer is no.`
`19:37.56`	`*** part/#asterisk hassler (n=hassler@r-corp.hcst.com)`
`19:38.06`	`tzafrir_laptop`	`HushPe, mercestes mixed zatel.conf and zapata.conf. channel => 1-6`
`19:38.33`	`*** join/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net)`
`19:38.36`	`[hC]`	`mercestes: :) Interesting answer. So you're saying, with some extra hacking, it can be done, but there is nothing to facilitate it out of the box?`
`19:38.38`	`tutt9876`	`hi`
`19:38.51`	`[hC]`	`mercestes: and by hacking, I mean hacking the code, not the dial plan. :)`
`19:38.53`	`tutt9876`	`A question about sound files`
`19:38.56`	`mercestes`	`[hC]: Yes to both cases. Another answer would be "not without modifying the source."`
`19:39.10`	`mercestes`	`[hC]: exactly the case.`
`19:39.24`	`tzafrir_laptop`	`[hC], in practical settings, you won't have catsex on your dialplan`
`19:39.25`	`tutt9876`	`may I Ask?`
`19:39.25`	`*** join/#asterisk reber (i=reber@gateway/tor/x-668dddf0b2cdcab8)`
`19:39.44`	`[hC]`	`tzafrir_laptop: in practical settings, i prefer not to have catsex at all!`
`19:40.04`	`tutt9876`	`my asterisk is looking for gsm files in /usr/share instead of /var/lib`
`19:40.09`	`mercestes`	`tutt9876: You may ask, my supplicant.`
`19:40.17`	`tutt9876`	`How can I change config?`
`19:40.48`	`tzafrir_laptop`	`tutt9876, apt-get install asterisk-sounds-extra`
`19:40.51`	`mercestes`	`tutt9876: Edi tasterisk.conf`
`19:40.53`	`potential`	`Anyone have some free time to help me get a server together?`
`19:40.56`	`mercestes`	`gah`
`19:41.07`	`mercestes`	`tutt9876: Edit asterisk.conf`
`19:41.31`	`tutt9876`	`no parameter in asteriks.conf for sound directory`
`19:41.51`	`mercestes`	`what does it say for astvarlibdir?`
`19:42.02`	`tzafrir_laptop`	`astdatadir is /usr/share/asterisk in your case`
`19:42.40`	`tzafrir_laptop`	`And a hird alternative: symlink`
`19:42.46`	`tutt9876`	`astvarlibdir => /var/lib/asterisk`
`19:43.01`	`mercestes`	`hmm.`
`19:43.12`	`tutt9876`	`I made a grep in asterisk.conf`
`19:43.15`	`mercestes`	`Try tzafrir's suggestion then.`
`19:43.38`	`tutt9876`	`tzafrir's and after installing the package?`
`19:43.43`	`mercestes`	`potential: How much is free time worth to you?`
`19:43.51`	`sevard`	`potential: check your messages.`
`19:44.47`	`*** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch)`
`19:46.16`	*** join/#asterisk ezw` (n=Ez@c66.203.210-59.clta.globetrotter.net)
`19:46.31`	`*** join/#asterisk djflux (n=djflux@mm.shermfin.com)`
`19:47.17`	`*** join/#asterisk topping (n=topping@204.152.96.50)`
`19:47.18`	`tutt9876`	`sorry I just apr-get for asterisk-sounds-extra but i still get some not found gsm files because of a wrong directory search`
`19:47.46`	`*** join/#asterisk Ritalin2 (n=none@c-68-47-199-178.hsd1.tn.comcast.net)`
`19:48.11`	`*** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)`
`19:48.50`	`Strom_C`	`tutt9876: edit asterisk.conf`
`19:49.03`	`EmleyMoor`	`When I get calls in over SIP, if the caller hangs up before I answer, it appears the calls "ring on" and then get diverted to the VoiceMail, which cuts out due to lack of audio. Why would the calls "ring on"?`
`19:49.04`	`tutt9876`	`And which parameter to change?`
`19:49.08`	`tzafrir_laptop`	`tutt9876, how can you tell?`
`19:49.32`	`Assid`	`hrmm.. anyone knoiw a sip client that can be used on a Sony Ericsson P990`
`19:49.34`	`tutt9876`	`Tzafrir: sorry?`
`19:49.44`	`tzafrir_laptop`	`when is asterisk.conf read to apply changes? only on restart of asterisk, right?`
`19:50.04`	`Strom_C`	`yeah`
`19:50.12`	`*** join/#asterisk jedir0x (n=bdilley@adsl-75-1-255-202.dsl.irvnca.sbcglobal.net)`
`19:50.23`	`*** part/#asterisk muh-die-kuh (n=hco@admin.labnine.de)`
`19:50.23`	`tutt9876`	`I did a relaod`
`19:51.23`	`jedir0x`	`Hi all. I'm an astrisk newbie (if i can even be considered that)... what i'd like to know is what all is needed (outside of astrisk) to setup a VoIP situation where people on regular land linds could make calls to it, and it could make calls outbound...`
`19:52.23`	`robin_sz`	`well, youd need some incoming lines to take the calls on`
`19:52.32`	`EmleyMoor`	`jedir0x: An incoming PSTN number, routed by a VoIP provider over SIP or IAX2, and outbound SIP or IAX2 connectivity with a provider that can router the calls to the PSTN for you`
`19:52.48`	`jedir0x`	`awesome, thanks :)`
`19:52.57`	`EmleyMoor`	`(I have all that and my PSTN line hooked in too)`
`19:53.13`	`jedir0x`	`hobby stuff?`
`19:53.30`	`jedir0x`	`cheaper than paying for service from somewhere else?`
`19:53.30`	`EmleyMoor`	`Well, semi-serious`
`19:54.07`	`*** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net)`
`19:54.27`	`EmleyMoor`	`This way we can both be on the phone at the same time, have our own numbers, our own voicemail... and calls route "cheapest available way"`
`19:54.44`	`tutt9876`	`some one ti help me with my directory problem?`
`19:54.50`	`tutt9876`	`to help`
`19:56.37`	`tzafrir_laptop`	`tutt9876, a. I believe you need to fully restart asterisk in order for asterisk.conf to e read`
`19:57.08`	`tzafrir_laptop`	`b. I asked how can you tell that the problem is with the base sounds path`
`19:57.41`	`tzafrir_laptop`	`asterisk -rx 'restart now'`
`19:57.56`	`tzafrir_laptop`	`in case you had no running calls and such nonsense`
`19:58.07`	`tutt9876`	`I just see that there was a asterisk.conf.dpkg-dist file with the wrong directory`
`19:58.26`	`tutt9876`	`do you know what is asterisk.conf.dpkg-dist ?`
`19:58.29`	`Ritalin2`	`who wants to help me figure out why hook flashing on my zap channel is causing it to hangup? :)`
`19:59.01`	`RoyK`	`tutt9876: just setup a cron job: 0 * * * * asterisk -rx "restart now"`
`19:59.17`	`mercestes`	`RoyK: Don't be an ass.`
`19:59.30`	`RoyK`	`rotfl`
`19:59.43`	`mercestes`	`Wasn't OpenPBX enough of a joke?`
`19:59.52`	`RoyK`	`works better for me`
`19:59.59`	`tutt9876`	`sorry for thaht stupid question : bu what should i do with asterisk.conf.dpkg-dist ?`
`20:00.04`	`tutt9876`	`but`
`20:00.13`	`mercestes`	`tutt9876: Never heard of asterisk.conf.dpkg-dist.`
`20:01.03`	`RoyK`	`it's what comes when you have an existing asterisk.conf and you upgrade or install from a dpkg`
`20:01.11`	`HushPe`	`i have a loop in my dial plan for some reason... background, wait, macro (which should dial), but it's looping the background message`
`20:01.17`	`RoyK`	`dpkg asks if you want to overwrite the original config, and you answer no, so it leaves that`
`20:01.25`	`HushPe`	`if i dial an extension it goes away and does that`
`20:02.38`	`*** join/#asterisk DavoFrom818 (n=Vito@cpe-76-173-56-41.socal.res.rr.com)`
`20:02.49`	`DavoFrom818`	`can anyone help me out im getting fast busy dialing out http://de.pastebin.ca/328400`
`20:02.55`	`HushPe`	`http://pastie.caboo.se/35659`
`20:03.10`	`HushPe`	`(waitexten does the same)`
`20:03.28`	`tutt9876`	`<tzafrir_laptop>: sorry I remove asterisk.conf.dpkg-dist and restart but no effect`
`20:03.54`	`RoyK`	`tutt9876: that file has no effect with asterisk`
`20:04.43`	`tutt9876`	`tzafrir: I remove a gsm file in some sub directory of /usr/share and asterisk didn't find the file it used before`
`20:05.24`	`tzafrir_laptop`	`asterisk.conf.dpkg-dist is the file dpkg leaves because it had a new version of a configuration file (asterisk.conf) but it didn't want to override the user's copy`
`20:05.34`	`tzafrir_laptop`	`This is the default policy with Debian`
`20:05.38`	`*** join/#asterisk PMantis (n=pmantis@66.251.89.34)`
`20:05.38`	`HushPe`	`got it! my sip extensions didn't quite exist yet ;)`
`20:06.16`	`tutt9876`	`ok if noone has a solution I will copy the missing files in the other directory`
`20:06.17`	`tzafrir_laptop`	`You get to change that on a per-file bais during an update (if the configuration file was actually chamged locally)`
`20:06.49`	`EmleyMoor`	`Has anyone got a number reachable through SIPbroker I could try?`
`20:06.54`	`sevard`	`now he changed it from a /say to a /me`
`20:06.58`	`tutt9876`	`tzfrir: didn't catch your answer`
`20:07.16`	`tzafrir_laptop`	`I see RoyK answered the same answer already`
`20:07.44`	`Ritalin2`	`I have an analog zap channel (FXS). Works fine with one call. If another call comes in and I hook flash this happens (it drops the first call) http://de.pastebin.ca/328415`
`20:08.12`	`tzafrir_laptop`	`tutt9876, what is your test to show where Asteris is taking sound files from?`
`20:08.33`	`tzafrir_laptop`	`Have you restarted Astersk since you last edited asterisk.conf?`
`20:08.44`	`tzafrir_laptop`	`can you pastebin your asterisk.conf?`
`20:09.10`	`tutt9876`	`yes restarted many times`
`20:09.14`	`Manfish`	`anyone here able to shed any light on a moh problem where the track restarts after each announcement in a queue instad of pausing and picking up where it left off`
`20:10.23`	`tutt9876`	`Ok I copy the missing files with a cp -Rp and it's ok: no idea why asterisk get the wrong directory..`
`20:10.30`	`tutt9876`	`thanks everyone`
`20:10.47`	`*** part/#asterisk tutt9876 (n=tut123@cvl92-2-82-228-144-230.fbx.proxad.net)`
`20:10.52`	`tzafrir_laptop`	`Ritalin2, are you sure that the flash wasnot interpeted as a hangup?`
`20:12.02`	`FuriousGeorge`	`my adventures with asterisk today seem to have something to do with this -> Notify answer on an owned channel?`
`20:12.07`	`Ritalin2`	`tzafrir: well it appears to be thought of that way. I tried using the flash button and tapping the hook as quickly as possible. both result in a hangup`
`20:12.08`	`FuriousGeorge`	`silly chan_sip`
`20:12.32`	`oej`	`it's not silly!!!!`
`20:12.43`	`FuriousGeorge`	`sorry oej`
`20:12.50`	`rene-`	`heh`
`20:12.51`	`oej`	`:-)`
`20:12.55`	`Ritalin2`	`<PROTECTED>`
`20:14.36`	`rene-`	`oej: why asterisk ## transfers can drop the transferer but SIP refer with replaces or without them keem em reported as busy/unavail`
`20:14.46`	`*** join/#asterisk s1gny\|wrk (n=s1gny@p549169FB.dip.t-dialin.net)`
`20:15.05`	`DavoFrom818`	`can anyone help me out im getting fast busy dialing out http://de.pastebin.ca/328400`
`20:15.06`	`*** part/#asterisk s1gny\|wrk (n=s1gny@p549169FB.dip.t-dialin.net)`
`20:15.06`	`oej`	`rene-: That is indeed a bug.`
`20:15.10`	`DavoFrom818`	`We're at 192.168.1.3 port 10078`
`20:15.13`	`DavoFrom818`	`is that my prob lem`
`20:15.20`	`oej`	`rene-: I need to test that. Thanks for reporting.`
`20:16.30`	`[hC]`	`oej: good on you for understanding rene's english there.. I couldnt make sense of it`
`20:16.44`	`oej`	`hC: It was SIPish!`
`20:16.57`	`[hC]`	`oej: Ha! :) I guess you take what you need out of it easily enough.. :P`
`20:17.47`	`rene-`	`[hC]: thats 1337 for you :P`
`20:18.14`	`PMantis`	`Anyone have pointers toget fax detection going in Asterisk 1.4 ? Zaptel detection seems to not work, and NVFaxDetect won't compile in 1.4`
`20:18.29`	`*** join/#asterisk saftsack (n=saftsack@193.218.17.212)`
`20:19.18`	`mercestes`	`PMantis: I suggest not using fax detection`
`20:19.27`	`De_Mon`	`sevard yeah, but it wasnt funny.`
`20:20.19`	`PMantis`	`mercestes, Not acceptable to my client. :( Their clients expect the 800 number to the queue to accept faxes as well.`
`20:20.35`	`De_Mon`	`any real answers to how to end processing a call thats in the h extension? i'm inclined to issue another Hangup()`
`20:20.37`	`mercestes`	`PMantis: It will never work with all faxes`
`20:20.39`	`PMantis`	`mercestes, All on a PRI, so Zap detection should work...`
`20:20.42`	`De_Mon`	`s/call/channel/`
`20:21.13`	`*** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com)`
`20:22.39`	`rene-`	`Pmantis: asterisk faxing has never been strong, why not use a did for faxing and send those calls out to something like hylafax and a modem or some fax HW`
`20:23.27`	`PMantis`	`rene-, I'm using Hylafax with iaxmodem - works great... but I just need to detect faxes to send them to the right destination.`
`20:24.28`	`Nugget`	`http://www.unitedmedia.com/comics/dilbert/archive/dilbert-20070125.html`
`20:24.55`	`*** join/#asterisk zotz (n=zotz@24.244.163.157)`
`20:25.42`	`mercestes`	`PMantis: Can't help you, I suggesting going back to 1.2.13 or 1.2.14 if this is production, 1.4 is not stable yet AFIAK. I still suggest giving up on the fax detect thing. It won't work with all faxes.`
`20:26.46`	`mercestes`	`PMantis: Your askin' for trouble there.`
`20:26.50`	`mercestes`	`Nugget: Very nice.`
`20:36.43`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`20:36.43`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`20:36.56`	`sahafeez`	`PMantis: you could just avoid the whole thing by having a set fax number and route by the inbound number dialed`
`20:37.39`	`PMantis`	`sahafeez, Yeah, that's not going to work well for them`
`20:38.11`	`sahafeez`	`i never understood that? what do they do just say oh, just call or fax me at the same number?`
`20:39.24`	`*** join/#asterisk zapx (n=zap@146-115-115-175.c3-0.lex-ubr2.sbo-lex.ma.cable.rcn.com)`
`20:39.39`	`zapx`	`I have been having a SIP registration problem... It stays on "Request Sent" (tried multiple SIP providers)... It's the DMZ, under SIP Registry it shows: proxy01.sipphone.com/1747 198.65.166.131 N 5060 Unmonitored. Any ideas?`
`20:43.24`	`PMantis`	`sahafeez, There's multple clients (of my client) that want fax and voice on the same number`
`20:45.16`	`*** join/#asterisk anthonyl (n=anthonyl@72.146.49.215)`
`20:47.27`	`ThoMe`	`hello`
`20:47.33`	`ThoMe`	`how i can "send text" to a display`
`20:47.43`	`ThoMe`	`or set the name from the source number?`
`20:47.56`	`bkruse`	`sendtext is kinda old and iffy`
`20:48.04`	`ThoMe`	`bkruse: hmm?`
`20:48.09`	`bkruse`	`works on some phones, some phones dont support it or have bad implementation`
`20:48.28`	`ThoMe`	`bkruse: kannst du deutsch? ;)`
`20:48.42`	`bkruse`	`..........`
`20:50.36`	`*** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa)`
`20:53.08`	`*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)`
`20:53.23`	`*** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir)`
`20:53.30`	`Zodiacal`	`anyone know if i really need an amplifier for a headset for a polycom 601?`
`20:53.38`	`bkruse`	`no`
`20:53.49`	`Zodiacal`	`also, which is better do you guys think, the tube headsets or the little plastic stub (noice canceling) ones?`
`20:54.03`	`Zodiacal`	`oh and is plantronics any good?`
`20:54.24`	`*** join/#asterisk has_many_questio (n=Deezzer@c-67-180-40-64.hsd1.ca.comcast.net)`
`20:54.44`	`Qwell[]`	`plantronics is good`
`20:55.26`	`rudholm`	`Qwell[]: /last CPSK`
`20:55.38`	`rudholm`	`he's at it again <sigh>`
`20:55.39`	`Qwell[]`	`no such command`
`20:55.54`	`mercestes`	`\last CPSK`
`20:55.56`	`mercestes`	`lol`
`20:56.20`	`mercestes`	`<CPSK> Hi, looking for help with Tenovis Integral E1 and TE110P.. msg me, thanx`
`20:56.30`	`bkruse`	`oh god.`
`20:56.54`	`bkruse`	`I needs help configuring the linux with the asterisk, kthx - russell`
`20:57.47`	`mercestes`	`bkruse: your kidding, right?`
`20:58.11`	`Zodiacal`	`if i get a headset that has an amp or a wireless base, can i plug it into the headset port on my phone and just press the headset button to answer or do i have to use the handset "Lifter"?`
`20:58.17`	`*** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com)`
`20:58.19`	`bkruse`	`mercestes: haha, yes`
`20:58.27`	`mercestes`	`bkruse: making sure..:)`
`20:59.06`	`mercestes`	`Zodiacal: No. And if you have polycom phones I suggest you hide the headset lifters and don't tell ANYONE you have any. and disevow any knowledge of them forever.`
`20:59.19`	`DaPrivateer`	`So my asterisk PBX has been running fine for 101 days. I reboot today cause someone at my office is a retard and unplugged the power cord, and now all the sudden im getting "Unable to open master device '/dev/zap/ctl'" when i try to ztcfg -vvv`
`20:59.31`	`DaPrivateer`	`this is on debeian (kernel 2.4) any one haev any ideas?`
`21:00.02`	`Zodiacal`	`mercestes seems like a weird device...`
`21:00.08`	`Aurs`	`DaPrivateer: check README.udev`
`21:00.25`	`mercestes`	`DaPrivateer: Check yoru permissions. And make sure /dev/zap/ctl even exists.`
`21:00.38`	`DaPrivateer`	`it does exist`
`21:00.45`	`DaPrivateer`	`the thing is nothing has changed on the box`
`21:00.58`	`mercestes`	`DaPrivateer: ....did you reboot?`
`21:01.09`	`*** join/#asterisk cyrk (n=cyrk@adsl-71-130-211-241.dsl.irvnca.pacbell.net)`
`21:01.11`	`DaPrivateer`	`of course`
`21:01.15`	`mercestes`	`Then things changed.`
`21:01.19`	`mercestes`	`Who owns /dev/zap/ctl?`
`21:01.22`	`anthonyl`	`are the driver actully loaded?`
`21:01.33`	`mercestes`	`root:root? root:dialup? root:asterisk? etc?`
`21:01.38`	`*** join/#asterisk clonaAway (n=clona@bjs2-dhcp111.studby.uio.no)`
`21:01.41`	`DaPrivateer`	`root:dialout`
`21:01.43`	`Jingles`	`don:trump`
`21:01.52`	`*** part/#asterisk clonaAway (n=clona@bjs2-dhcp111.studby.uio.no)`
`21:01.52`	`Jingles`	`just a guess.`
`21:01.55`	`mercestes`	`DaPrivateer: What user are you trying to use to run *?`
`21:02.01`	`DaPrivateer`	`root`
`21:02.05`	`mercestes`	`are you sure?`
`21:02.10`	`DaPrivateer`	`yes`
`21:02.23`	`mercestes`	`hwo are you running *?`
`21:02.51`	`DaPrivateer`	`im not even getting to that point mercestes`
`21:03.00`	`DaPrivateer`	`im still trying to ztcfg -vvv`
`21:04.07`	`mercestes`	`do a modprobe zaptel and try again`
`21:04.28`	`DaPrivateer`	`says it can't locate the module, even though the module is there`
`21:04.40`	`mercestes`	`ew`
`21:05.08`	`DaPrivateer`	`http://pastebin.com/867412`
`21:05.21`	`Aurs`	`DaPrivateer: less <zaptel source dir>/README.udev`
`21:06.29`	`mercestes`	`lol. I would go with Aurs' suggestion. There may be something important in there.`
`21:07.08`	`Aurs`	`;)`
`21:07.17`	`DaPrivateer`	`im not running udev`
`21:07.22`	`DaPrivateer`	`afaik`
`21:07.27`	`cyrk`	`can anyone help me with a cisco 7960? It boots up and says tft timeout..I can't get into the menu or anything..tried resetting numerous times`
`21:07.47`	`Aurs`	`/dev/zap/ctl sounds like udev`
`21:08.20`	`Aurs`	`or... hmm`
`21:08.30`	`DaPrivateer`	`im running kernel 2.4.27`
`21:08.37`	`DaPrivateer`	`doesnt udev need 2.6?`
`21:08.37`	`Aurs`	`if you were running udev, the devices shouldn't be there before loading the module I gues`
`21:09.45`	`markit`	`is it possible to run asterisk under a VM, like with KVM? or there are performance/responsiveness issues too bad?`
`21:09.55`	`Qwell[]`	`markit: it's possible, but not recommended`
`21:10.24`	`Qwell[]`	`some people have had good luck with it, others have seen horrible results`
`21:10.42`	`markit`	`Qwell[]: any doc about the issues involved? would be great for me`
`21:10.45`	`DaPrivateer`	`Aurs or mercestes any other thoughts?`
`21:11.00`	`bkruse`	`markit: asteriskNOW has VM images already built`
`21:11.43`	`Aurs`	`DaPrivateer: you're correct. it needs 2.6`
`21:12.01`	`markit`	`Qwell[]: so the only real recommanded way is a "stand alone", dedicated pc?`
`21:12.03`	`Aurs`	`so you're probably not running udev ;)`
`21:12.07`	`DaPrivateer`	`sigh my boss is screaming at me right now, and i have no idea what to tell him`
`21:12.27`	`qdk`	`what does an asterisk in graceful restart/stop mode signal ZAP/SIP/IAX channels, so that they know its "offline"? and does it signal all those 3 channels according to specifications?`
`21:12.41`	`*** join/#asterisk Fausted (n=dfas@68.Red-213-98-224.dynamicIP.rima-tde.net)`
`21:12.43`	`Aurs`	`DaPrivateer: tell him to quit unplugging your servers`
`21:12.44`	`mercestes`	`DaPrivateer: Nice. You work for Cytel?`
`21:12.44`	`markit`	`Qwell[]: or havin that pc that does also samba file server, KDE remote X support, etc is just fine?`
`21:12.46`	`Fausted`	`Hi`
`21:13.16`	`mercestes`	`DaPrivateer: well, first, what distro are you on?`
`21:13.18`	`DaPrivateer`	`mercestes - nope`
`21:13.24`	`DaPrivateer`	`Debian 3`
`21:13.28`	`Fausted`	`I have a littler question about the election for asterisk`
`21:13.33`	`Fausted`	`I bought a x100p CARD`
`21:13.45`	`Qwell[]`	`election?`
`21:14.05`	`Fausted`	`yes`
`21:14.10`	`mercestes`	`sighs Couldn't be gentoo.`
`21:14.23`	`mercestes`	`It can't find the module even though the module is there?`
`21:14.24`	`Fausted`	`my question is what is the better distribution for asterisk ?`
`21:14.42`	`Fausted`	`I tried asterisknow but some things doesn't work`
`21:14.42`	`*** join/#asterisk PMantis (n=pmantis@66.251.89.34)`
`21:14.43`	`qdk`	`Fausted: whatever dist. you know best.`
`21:14.58`	`*** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir)`
`21:15.16`	`DaPrivateer`	`its def. there`
`21:15.25`	`qdk`	`Fausted: you can put my vote on Debian, if you care.`
`21:15.27`	`mercestes`	`are you in panic mode?`
`21:15.33`	`markit`	`Fausted: well, I have some issues with debian and kernel module compilation (i.e. zaptel), I solved using a kernel from the new, debian based, sidux distro`
`21:15.41`	`mercestes`	`Fausted: I vote gentoo`
`21:15.58`	`PMantis`	`Can anyone help with nv_fax_detect compile errors? http://paste.biz/paste-395.html`
`21:16.17`	`Fausted`	`ok`
`21:16.17`	`DaPrivateer`	`mercestes - luckily im off site right now`
`21:16.24`	`DaPrivateer`	`so i just dont answer my cell :-p`
`21:16.28`	`qdk`	`markit: i bet it wasnt debian fault.`
`21:16.29`	`Fausted`	`and then what is the step for set pstn with sip ?`
`21:16.36`	`Fausted`	`because I tried with asterisknow`
`21:16.45`	`Fausted`	`but I can't make work it`
`21:16.46`	`qdk`	`Fausted: read a guide/howto.`
`21:16.50`	`mercestes`	`DaPrivateer: well, if you wanna risk blowing up the filesystem (which could ahve been done already).`
`21:17.01`	`Fausted`	`qdk Yes I read, but don't work :(`
`21:17.09`	`markit`	`qdk: asterisk guys think so, instead`
`21:17.23`	`mercestes`	`DaPrivateer: cd /usr/src/linux then make modules_prepare && make modules_install assuming that works in Debian linux. (should).`
`21:17.32`	`*** join/#asterisk PupenoR (n=pupeno@200.123.183.91)`
`21:17.38`	`mercestes`	`DaPrivateer: make a backup of zaptel and zapata.conf and give zaptel a nice recompile too. Then try the modprobe zaptel`
`21:18.22`	`DaPrivateer`	`blowing up the file system probably isnt a good idea`
`21:18.23`	`qdk`	`markit: ok, I dont care, so lets say that the "asterisk guys" are right.`
`21:18.37`	`DaPrivateer`	`mercestes - im using versions from apt-get`
`21:18.45`	`DavoFrom818`	`hi can anyone tell me what this means? Jan 25 13:16:03 NOTICE[5200] chan_sip.c: Failed to authenticate on INVITE to '"ITC INC" ;tag=as701112da'`
`21:18.46`	`DaPrivateer`	`i could never actually get it to compile on the box`
`21:18.47`	`Fausted`	`and the better GUI for asterisk ?`
`21:18.49`	`Fausted`	`freepbx ?`
`21:18.49`	`*** join/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca)`
`21:19.01`	`*** part/#asterisk Telamon (i=telamon@blk-137-96-217.eastlink.ca)`
`21:19.08`	`qdk`	`Fausted: then ask questions to what you have done wrong instead of expecting a spoon feeding.`
`21:19.23`	`DavoFrom818`	`i asked how come no one will answer`
`21:19.37`	`slima`	`asterisk 1.4 need root account for install, and working?`
`21:19.42`	`Fausted`	`Ok sorry qdk`
`21:20.08`	`mercestes`	`Fausted: I suggest against any GUI and instead, suggest using vanilla asterisk.`
`21:20.33`	`DaPrivateer`	`mercestes - intersting: http://pastebin.com/867431`
`21:20.38`	`mercestes`	`Fausted: If you want a GUI consider using "RTA" and put some of the config files into a Mysql database and write your own interface.`
`21:20.47`	`qdk`	`slima: you could probably make it work without, but will somewhat limited functionality.`
`21:21.04`	`Hmmhesays`	`bah skype is so damn easy`
`21:21.21`	`slima`	`qdk: mhm, thx.`
`21:21.48`	`anonymouz666`	`for the first time I got that: The previous reload command didn't finish yet`
`21:21.49`	`anonymouz666`	`lol`
`21:21.51`	`hads`	`If you want your phone system reliant on a database that is.`
`21:22.13`	`*** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it)`
`21:22.36`	`DaPrivateer`	`you could also right a script to generate config files from the database, either cronned or when you click something on the custom web interface hads`
`21:22.52`	`DaPrivateer`	`and by right i mean write`
`21:23.08`	`mercestes`	`hads: =/ of all the failure points...I would think a DB is the least worrysome.`
`21:23.13`	`hads`	`You could. or you could just write the config files.`
`21:23.28`	`mercestes`	`I am pro config file`
`21:23.30`	`hads`	`More failure points == more failure`
`21:23.46`	`mercestes`	`Cluser mysql.`
`21:24.10`	`hads`	`or you could just write the config files :)`
`21:24.20`	`Aurs`	`or use traditional PSTN`
`21:24.23`	`mercestes`	`DaPrivateer: Oh crap`
`21:24.28`	`DaPrivateer`	`mercestes did my pastebin mean anything to you?`
`21:24.30`	`Aurs`	`and dont bother with asterisk at all`
`21:24.42`	`Aurs`	`then you have very few failure points :)`
`21:24.58`	`hads`	`True`
`21:24.59`	`sweeper`	`is there a list of feature bounties somewhere?`
`21:25.13`	`mercestes`	`DaPrivateer: Yea, your zaptel isn' tcompling..:P`
`21:25.13`	`Aurs`	`sweeper: yes, there is one on voip-info.org`
`21:26.06`	`mercestes`	`DaPrivateer: Did you run a chdsk on this box? chkdsk. fixdisk...whatever it's called?`
`21:26.09`	`Aurs`	`DaPrivateer: did you say that you used apt-get to install asterisk/zaptel?`
`21:26.22`	`mercestes`	`DaPrivateer: I don't tend to randomly manage to unplug dual powersupply servers so ..yea`
`21:26.31`	`DaPrivateer`	`yes; i have never been able to get it to compile successfully on debian`
`21:26.54`	`hads`	`I compile zaptel on Debain all the time so I don' think it has anything to do with that.`
`21:27.14`	`DaPrivateer`	`mercestes - they unplugged it cause none of the phones could connect, all of the sudden. of course, now i won't know what caused that cause they unplugged it`
`21:27.40`	`DaPrivateer`	`my boss refuses to lock the area where are servers are cause it would cost too much to put up a f*ing door, apparently`
`21:27.41`	`hads`	`So put it in a locked room`
`21:27.49`	`mercestes`	`DaPrivateer: so give it a nice apt-get zaptel then.`
`21:27.50`	`hads`	`or not`
`21:27.58`	`*** part/#asterisk trevarthan (n=trevarth@c-71-59-48-26.hsd1.ga.comcast.net)`
`21:28.03`	`*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)`
`21:28.04`	`DaPrivateer`	`mercestes - i did lol`
`21:28.08`	`mercestes`	`DaPrivateer: Ask him how much revenu ehe's loosing on this particular.....debauchrey.`
`21:28.13`	`DaPrivateer`	`i tried that long before bothering you guys :-p`
`21:28.18`	`mercestes`	`DaPrivateer: Anythign on fsck ?`
`21:28.20`	`Hmmhesays`	`the skype guys are offering a year of unlimited calling for 14.95`
`21:28.23`	`Hmmhesays`	`crazy`
`21:29.23`	`DaPrivateer`	`mercestes - says its clean`
`21:29.52`	`Strom_C`	`Here's a really dumb question - which ATAs support T.38?`
`21:30.06`	`rudholm`	`I love my doorphone :)`
`21:30.22`	`Strom_C`	`it's a pretty sweet doorphone, rudholm :)`
`21:30.29`	`rudholm`	`the new pool service showed up just now and I answered from my office 20 miles away`
`21:30.39`	`rudholm`	`told him he could just let himself in the side gate`
`21:30.39`	`mercestes`	`DaPrivateer: =/ And apt-get is failing.`
`21:30.48`	`*** part/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com)`
`21:30.58`	`Strom_C`	`rudholm: that is so kickass :)`
`21:31.02`	`DaPrivateer`	`no, apt-get says it worked fine`
`21:31.17`	`mercestes`	`DaPrivateer: And apt-get compiles nicely?`
`21:31.29`	`DaPrivateer`	`apt-get doesnt compile; it downloads binaries`
`21:31.30`	`hads`	`?`
`21:31.34`	`mercestes`	`oh.`
`21:31.51`	`mercestes`	`so you get it, but then it won't compile. ..nice Are you getting the right version?`
`21:32.10`	`hads`	`?`
`21:32.20`	`DaPrivateer`	`afaik, yes`
`21:32.27`	`DaPrivateer`	`it doesnt really give me an option`
`21:32.29`	`mercestes`	`hads: ?`
`21:32.40`	`DaPrivateer`	`what still kills me is that it worked before, and now doesnt. its confusing as hell`
`21:32.50`	`DaPrivateer`	`i considered the possibility that the card died, but it still appears to be software`
`21:32.58`	`hads`	`"< DaPrivateer> apt-get doesnt compile; it downloads binaries" "< mercestes> so you get it, but then it won't compile."`
`21:33.00`	`mercestes`	`DaPrivateer: I agree.`
`21:33.17`	`mercestes`	`hads: oh...downloads binaries.`
`21:33.24`	`mercestes`	`....cries`
`21:33.33`	`mercestes`	`ppl and their binaries. Source is tasty`
`21:33.44`	`mercestes`	`DaPrivateer: Ok, so you get new binaries and you can't modprobe it still?`
`21:33.50`	`DaPrivateer`	`correct`
`21:33.58`	`DaPrivateer`	`and i tried to insmod the new binaries and it still didnt work`
`21:34.03`	`DaPrivateer`	`alas, im gonna try upgrading to kernel 2.6`
`21:34.17`	`DaPrivateer`	`ive been meaning to do it for a while`
`21:34.26`	`mercestes`	`Now would be a good time. :0`
`21:34.44`	`DaPrivateer`	`im downloading the kernel source for 2.6.19 right now :-p`
`21:35.20`	`mercestes`	`see..this is why I vote gentoo`
`21:35.23`	`DaPrivateer`	`meh`
`21:35.25`	`DaPrivateer`	`i prefer freebsd`
`21:35.28`	`*** join/#asterisk hardwire (n=hardwire@rdbck-2645.wasilla.mtaonline.net)`
`21:35.30`	`hardwire`	`mofos`
`21:35.32`	`DaPrivateer`	`but asterisk doesnt really like freebsd`
`21:35.40`	`sweeper`	`yea D:`
`21:35.48`	`hardwire`	`JT: was I bugging you the other day?`
`21:35.53`	`hardwire`	`I am sure of it`
`21:36.00`	`rudholm`	`if you prefer FBSD, why are you downloading (linux) kernel 2.6?`
`21:36.10`	`rudholm`	`:)`
`21:36.15`	`mercestes`	`rudholm: LOL`
`21:36.16`	`DaPrivateer`	`rudholm - [16:35] <DaPrivateer> but asterisk doesnt really like freebsd`
`21:36.21`	`hads`	`DaPrivateer: You will just get yourself into more trouble mixing source installs and debs`
`21:36.42`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`21:36.43`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`21:36.46`	`DaPrivateer`	`hads - im hoping i will be able to compile it under 2.6`
`21:36.53`	`rudholm`	`but you said you prefer FBSD in response to a gentoo recommendation`
`21:36.55`	`rudholm`	`which doesn't make sense`
`21:37.04`	`rudholm`	`since FBSD isn't an option`
`21:37.14`	`hads`	`I'll leave you all to it.`
`21:37.15`	`rudholm`	`'sall I'm sayin`
`21:37.23`	`mercestes`	`hads: If you have an answer........`
`21:37.25`	`DaPrivateer`	`mercestes was knocking my choice of debian, so i was countering that i dont like it much either`
`21:37.32`	`rudholm`	`ah`
`21:37.50`	`rudholm`	`I'm running Asterisk on a couple of gentoo systems`
`21:38.18`	`mercestes`	`DaPrivateer: Yea, "emerge zaptel libpri asterisk asterisk-sounds asterisk-addons" is inexcusably difficult..;)`
`21:38.19`	`rudholm`	`the package database is generally pretty current (the Asterisk package maintainer is generally quick)`
`21:38.39`	`mercestes`	`rudholm: stkn is my hero.`
`21:42.57`	`*** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net)`
`21:43.13`	`BB\|AtWork`	`how does one start on demand call recording?`
`21:43.21`	`Aurs`	`anyone using polycom buddy list in asterisk here?`
`21:43.40`	`*** join/#asterisk bkw_ (n=brian@adsl-70-143-45-86.dsl.tul2ok.sbcglobal.net)`
`21:43.59`	`*** part/#asterisk cyrk (n=cyrk@adsl-71-130-211-241.dsl.irvnca.pacbell.net)`
`21:46.06`	`DaPrivateer`	`configuration complete; compile run has begun`
`21:46.46`	`DaPrivateer`	`at least this things 3 Ghz; i had to compile 2.6 the other day on a 700 Mhz system ... sucked`
`21:48.40`	`*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)`
`21:51.18`	`HushPe`	`i have my first working 'real' asterisk pbx`
`21:51.40`	`shido6`	`congrats`
`21:51.56`	`De_Mon`	`huzah`
`21:52.05`	`HushPe`	`cheers :) dial plan still still need changing as we add a few more phones, and another TDM400P card`
`21:52.52`	`HushPe`	`but it's functional welcome message, moh, extensions, outgoing dialing :) haven't got DISA working just yet, but i'll get there, oh, and emailed voicemail! << so love that feature!`
`21:54.44`	`HushPe`	`next one will be the caller id database :) i should be able to get that working with mysql :) and send names through to the phone`
`21:54.57`	`rudholm`	`mercestes: even making the tweaks I had to make to the zaptel driver was easy with gentoo. one simply has to know how to use the ebuild command so that the various steps in an emerge can be done separately`
`21:55.15`	`rudholm`	`mercestes: or one could simply write a patch and put it in thier portage overlay directory`
`21:55.33`	`mercestes`	`compile from source is a beautiful thing.`
`21:56.11`	`rudholm`	`I had to make a couple of changes (to get dial pulse to work`
`21:56.13`	`rudholm`	`)`
`21:56.38`	`rudholm`	`the de-bounce timings weren't very good, so dial pulse interpretation didn't work`
`21:56.57`	`rudholm`	`it's a pretty well-known issue that I expect to be fixed in an upcoming release`
`21:57.29`	`*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)`
`21:57.37`	`rudholm`	`does anyone know if an IAXy does CPC on both inbound and outbound calls?`
`21:57.48`	`Strom_C`	`rudholm: let me go test`
`21:57.50`	`Strom_C`	`one moment please :)`
`21:57.56`	`[hC]`	`anyone here use commpartners for SIP termination?`
`21:58.00`	`rudholm`	`Strom_C: thanks`
`21:58.01`	`zapx`	`i have a very strange problem, i am unable to get any sip trunks to register.. if i tcpdump on the machine i see the udp packets being sent out on 5060.. if i run tcpdump on a machine that sees all packets on the network.. i see that the packets dont actually leave the asterisk server... if i 'telnet proxy01.sipphone.com 5060' i do see the packets in the lan tcpdump.. any ideas?`
`21:58.48`	`rudholm`	`Strom_C: as you might have guessed, I need more FSX channels and if the IAXy does CPC on both in and outbound, I could use it for my doorphone`
`21:59.41`	`rudholm`	`Strom_C: the Sipuras not doing CPC on outbound calls made it not suitable for the doorphone, and despite the myriad settings on an SPA, there isn't a way to turn it on.`
`21:59.51`	`*** join/#asterisk MonkeyHugs (n=jojo@63.149.122.93)`
`22:02.24`	`pjz`	`so I've had a feature request that I don't even know what to call to be able to see if it's possible`
`22:02.37`	`Strom_C`	`rudholm: the answer is "yes"`
`22:02.46`	`Strom_C`	`pjz: describe it`
`22:02.55`	`rudholm`	`Strom_C: oh cool`
`22:02.56`	`pjz`	`Joe wants to be able to call Bob, and, if Bob is on the phone, to see 'Bob is on the phone'`
`22:03.02`	`DavoFrom818`	`Jan 25 13:57:17 NOTICE[7607] chan_sip.c: Failed to authenticate on INVITE to '"ITC INC" ;tag=as55454e68' <<<<< Does this mean my password or username is wrong?`
`22:03.14`	`Strom_C`	`pjz: "see"? are they using videophones?`
`22:03.19`	`pjz`	`er, to see 'Bob is on the phone' displayed on his (Joe's) phone`
`22:03.19`	`rudholm`	`Strom_C: not surprising, since the TDM400 card does it both ways as well.`
`22:03.35`	`Strom_C`	`pjz: they want to see this /after/ they dial the call?`
`22:03.44`	`pjz`	`Strom_C: yes`
`22:04.04`	`Strom_C`	`that makes no sense`
`22:04.19`	`Strom_C`	`either you can set it to display status on the phone before the call is placed`
`22:04.25`	`Strom_C`	`or you can play a recording after they place the call`
`22:04.57`	`rene-`	`~seen oej`
`22:05.38`	`jbot`	`oej <n=olle@apollo.webway.se> was last seen on IRC in channel #asterisk, 1h 48m 54s ago, saying: 'hC: It was SIPish!'.`
`22:05.38`	`*** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net)`
`22:05.38`	`*** join/#asterisk Lokiji (n=Lokiji@ip-89-102-178-195.karneval.cz)`
`22:05.39`	`pjz`	`so Bob (on an internal extension) is talking to Carol (anywhere) on the phone. Joe calls Bob. Joe wants his phone to display 'Bob is on the phone', as well as have it ring Bob`
`22:05.39`	`rudholm`	`I really wish the IAXy was < 90$`
`22:06.01`	`pjz`	`Strom_C: does that make more sense?`
`22:06.05`	`Strom_C`	`pjz: that is going to be highly dependent on your station equipment, and I don't even know if that's possible`
`22:06.17`	`pjz`	`Strom_C: we've got all Polycom 501s`
`22:06.21`	`pjz`	`Strom_C: me either`
`22:06.38`	`Strom_C`	`better to design a solution that isn't so dependent on station equipment - play a short tone before alerting the called party`
`22:07.37`	`Strom_C`	`rudholm: do you want an iaxy instead of the $50 I owe you for the sofa?`
`22:08.53`	`DaPrivateer`	`mercestes still around?`
`22:09.00`	`*** join/#asterisk Fausted (n=dfas@68.Red-213-98-224.dynamicIP.rima-tde.net)`
`22:09.06`	`mercestes`	`no`
`22:09.12`	`DaPrivateer`	`hehe`
`22:09.18`	`mercestes`	`lol.. .Yea, what's up?`
`22:09.19`	`DaPrivateer`	`well, zaptel compiles now, but still doesnt work`
`22:09.40`	`Jingles`	`have you loaded the wctdm module first?`
`22:09.56`	`Jingles`	`wait. that's only on my system, since I'm using a TDM card.`
`22:09.57`	`Jingles`	`:P`
`22:10.22`	`Jingles`	`something about having to use ztdummy if you're not using a TDM card comes to mind, though.`
`22:12.00`	`Lokiji`	`hello could someone help me? how can i configure more than 1 sip trunk?i mean "register....." and i would like to chose the context for each sip incomink trunk`
`22:12.39`	`*** join/#asterisk netstatic (i=netstati@neptune.negativeblue.com)`
`22:12.48`	`netstatic`	`what is the command to drop a SIP channel in the * cli?`
`22:13.02`	`Strom_C`	`netstatic: soft hangup (channel)`
`22:13.19`	`netstatic`	`Strom_C: i can't seem to get the channel part correct`
`22:13.28`	`Strom_C`	`how about using tab-complete`
`22:13.37`	`Strom_C`	`soft hangup <tab>`
`22:14.03`	`netstatic`	`ahh, awesome`
`22:14.07`	`netstatic`	`thank you very much`
`22:14.46`	`*** part/#asterisk netstatic (i=netstati@neptune.negativeblue.com)`
`22:15.16`	`DaPrivateer`	`so im assuming i should run 1.2.14, cause im betting there are a lot of changes in 1.4?`
`22:15.25`	`DaPrivateer`	`cd ..`
`22:15.27`	`DaPrivateer`	`oops`
`22:15.33`	`Jingles`	`1.4 is not a stable release, apparently.`
`22:15.42`	`Jingles`	`which explains all the trouble I was having with it on my dev box.`
`22:15.49`	`Strom_C`	`it's 1.4.0`
`22:16.01`	`Strom_C`	`when was the last time you had a completely .0 release of ANYTHING?`
`22:16.07`	`Strom_C`	`er, completely stable`
`22:16.08`	`Jingles`	`fair enough.`
`22:16.10`	`pjz`	`Strom_C: the point is to alert the caller, not the callee`
`22:16.23`	`Strom_C`	`pjz: yes`
`22:16.34`	`file`	`if you do have issues try the latest 1.4 from SVN, and if it still happens then file a bug on Mantis`
`22:16.46`	`file`	`and we will try to fix it... but keep in mind we are not gods`
`22:17.02`	`Strom_C`	`thats why I said "play a tone before alerting the called party" which implies "play a tone to the calling party before beginning to ring the called party's telephone set"`
`22:17.13`	`*** join/#asterisk matt_ (n=matt@82-33-68-44.cable.ubr01.trow.blueyonder.co.uk)`
`22:17.34`	`pjz`	`Strom_C: ah, I see. I misunderstood who you were playing the tone to.`
`22:17.43`	`Strom_C`	`alerting == ringing`
`22:17.44`	`pjz`	`Strom_C: so how would I do that?`
`22:18.05`	`Strom_C`	`pjz: easily`
`22:18.19`	`Strom_C`	`do your status check and conditional branch before you execute Dial()`
`22:18.37`	`*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)`
`22:18.56`	`pjz`	`Strom_C: ah, I guess I don't know how to do the status check. conditional branches I've done, but not that kind of status check`
`22:19.11`	`Strom_C`	`try chanisavail()`
`22:20.31`	`*** join/#asterisk matt_ (n=matt@82-33-68-44.cable.ubr01.trow.blueyonder.co.uk)`
`22:21.02`	`*** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2)`
`22:21.14`	`matt_`	`hello :)`
`22:21.19`	`Strom_C`	`balls`
`22:21.28`	`matt_`	`i have a asterisk setup and everything it working fine`
`22:21.51`	`matt_`	`but i want to add a new phone, but this phone is routed to my asterisk box (not natted)`
`22:22.10`	`matt_`	`and i get one way audio unless i change the localnet= line in sip.conf`
`22:22.19`	`matt_`	`is it possiable to have 2 subnets on the localnet line?`
`22:22.54`	`mercestes`	`DaPrivateer: zOmg? You went from 1.2.14 to 1.4?`
`22:23.24`	`bkruse`	`~zomg`
`22:23.28`	`jbot`	`from memory, zomg is making fun of internet people`
`22:23.36`	`bkruse`	`lame.`
`22:23.39`	`bkruse`	`someone changed the good oen.`
`22:23.40`	`bkruse`	`one*`
`22:23.46`	`sivana`	`~bkruse`
`22:23.53`	`mercestes`	`~mercestes`
`22:23.54`	`jbot`	`methinks mercestes is is the almighty dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existance. Mercestes is also my Evil Twin!`
`22:24.05`	`Strom_C`	`some idiot misspelled existence`
`22:24.11`	`mercestes`	`What?`
`22:24.14`	`mercestes`	`that RETARD!`
`22:24.22`	`mercestes`	`>.>`
`22:24.27`	`sivana`	`~sivana`
`22:24.29`	`jbot`	`you are probably not exactly the sharpest tool in the shed`
`22:24.32`	`sivana`	`bah`
`22:25.16`	`mercestes`	`jbot, no, mercestes is the almight dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existence. Mercestes is also my evil twin!`
`22:25.18`	`jbot`	`mercestes: okay`
`22:25.38`	`mercestes`	`dum people, can't even spell existence.`
`22:25.43`	`sivana`	`jbot, no, sivana really is the sharpest tool in the shed`
`22:25.45`	`jbot`	`okay, sivana`
`22:26.08`	`Strom_C`	`~strom`
`22:26.10`	`jbot`	`strom is probably the coolest #asterisk lurker`
`22:26.12`	`Qwell[]`	`is a nub`
`22:26.16`	`Strom_C`	`~strom_c`
`22:26.18`	`jbot`	`you are probably just some nub`
`22:26.18`	`Qwell[]`	`stupid bot`
`22:26.22`	`Qwell[]`	`pwned`
`22:26.24`	`sivana`	`heh`
`22:26.42`	`mercestes`	`lol`
`22:26.44`	`Strom_C`	`hey, i am a proud card-carrying member of the digium nub club`
`22:26.58`	`Aurs`	`mercestes: dum?`
`22:27.32`	`sivana`	`mercestes: you retard.. you can't even spell almighty`
`22:27.44`	`sivana`	`hehe`
`22:27.51`	`mercestes`	`Ah crap`
`22:27.56`	`mercestes`	`jbot, no, mercestes is the almighty dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existence. Mercestes is also my evil twin!`
`22:27.58`	`jbot`	`mercestes: okay`
`22:28.06`	`mercestes`	`I hope I got it right this time.`
`22:28.17`	`sivana`	`hehe`
`22:28.27`	`*** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com)`
`22:28.39`	`mercestes`	`~twisted`
`22:28.41`	`jbot`	`hmm... twisted is twisted@indigent-networks.com, but you can paypal him at toastido@toastido.net`
`22:28.48`	`mercestes`	`boring.`
`22:28.49`	`sivana`	`haha`
`22:29.02`	`matt_`	`~matty`
`22:29.26`	`[TK]D-Fender`	`matt_ : No, you cannot put multiple subnets on a "localnet" line. You can however have multiple "localnet" lines...`
`22:29.45`	`matt_`	`[TK]D-Fender, cheers i have it working now :)`
`22:30.05`	`matt_`	`had to do a stop now and restart it wouldn't work from a module reload for some reason`
`22:32.27`	`mercestes`	`Strom_C: You are now hired as my grammar checker.`
`22:32.51`	`Strom_C`	`woot!`
`22:34.19`	`[hC]`	`aaah.. chan_sccp you kill me.`
`22:34.22`	`[hC]`	`you kill me i say`
`22:34.29`	`[hC]`	`i wonder if 7970+SIP is usable yet.`
`22:34.39`	`[hC]`	`Last time i tried, when the firmware just came out, i couldnt even get it to register`
`22:36.43`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`22:36.43`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`22:37.52`	`*** join/#asterisk sb_mx (n=sb_mx@200.78.229.18)`
`22:37.55`	`DavoFrom818`	`the register string is for incoming calls?`
`22:40.03`	`EmleyMoor`	`DavoFrom818: In general, yes, though if you have a section for the given number, any settings in it override [general]`
`22:41.04`	`DavoFrom818`	`question please`
`22:41.08`	`DavoFrom818`	`app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)`
`22:41.13`	`DavoFrom818`	`what does this mean`
`22:41.22`	`*** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net)`
`22:41.50`	`robin_sz`	`it means ,there is no route to the destination ...`
`22:41.50`	`elriah`	`Hi all. What's a good network switch to use with asterisk? one that will prioritize voice?`
`22:41.50`	`robin_sz`	`network error I guess`
`22:42.00`	`robin_sz`	`try doing a traceroute to the IP its trying to contact`
`22:42.43`	`robin_sz`	`elriah, thats a more complex question`
`22:42.54`	`DavoFrom818`	`any answers for mine?`
`22:43.12`	`robin_sz`	`elriah, you should probably put your phones on a separate vlan`
`22:43.21`	`EmleyMoor`	`DavoFrom818: Do you know which server it is trying to connect to?`
`22:43.43`	`DavoFrom818`	`yes`
`22:43.55`	`robin_sz`	`can you run traceroute to it?`
`22:43.57`	`CunningPike`	`elriah: We've had good success with Cisco 3750s`
`22:44.01`	`EmleyMoor`	`OK - can you traceroute from your * box to it?`
`22:44.12`	`DavoFrom818`	`yes i can`
`22:44.35`	`EmleyMoor`	`Do you have any other SIP providers defined?`
`22:44.50`	`DavoFrom818`	`yes just one but its no in the outbout route`
`22:45.34`	`EmleyMoor`	`Damn - still, does it work?`
`22:45.54`	`DavoFrom818`	`no i dont want to use the other one for outbound`
`22:46.31`	`EmleyMoor`	`But does it work? (whether you WANT to use it or not, this is a technical question)`
`22:46.38`	`DavoFrom818`	`i have a innomedia now im in the config of that device how do i know how to create a trunk with those settings`
`22:48.58`	`*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)`
`22:49.07`	`robin_sz`	`elriah, in a nutshell, you can either buy expnsive routers and switches with "voice" capability andd hope they do the right thing, or just use standard gear and set up vlans and assign vlan priorities etc`
`22:50.00`	`*** join/#asterisk UVSoft (n=UVSoft@c7204-ge2-500.etelecom.ru)`
`22:52.12`	`CunningPike`	`elriah: Or just have a fast enough network that it doesn't matter :)`
`22:53.22`	`*** join/#asterisk Ritalin2 (n=dave@74-34-103-241.dsl1.pwll.tn.frontiernet.net)`
`22:53.26`	`Ritalin2`	`what it is fellas?`
`22:53.50`	`EmleyMoor`	`What is what, which fellas?`
`22:54.10`	`Ritalin2`	`everyone. just asking how everyone is doing`
`22:54.18`	`Ritalin2`	`ladies as well`
`22:54.22`	`elriah`	`Thanks, all.`
`22:54.35`	`HushPe`	`is it possible to transfer from a zap ('real phone') by flashing the hook or something?`
`22:55.04`	`Ritalin2`	`HushPe: it should be. if you set threewaycalling=yes in zapata.conf`
`22:55.06`	`EmleyMoor`	`HushPe: Yes. Might need enabling but even works from my rotary`
`22:55.28`	`HushPe`	`nice one`
`22:55.35`	`HushPe`	`i'll add that and give it a shot!`
`22:55.37`	`Ritalin2`	`or blind transfer with # if that's enabled`
`22:55.41`	`EmleyMoor`	`(British TBR doesn't work, at least if you recompile to squelch the debounce so rotary phones work)`
`22:57.51`	`*** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it)`
`22:58.54`	`Ritalin2`	`HushPe: you also need to set transfer=yes`
`22:59.05`	`HushPe`	`Ritalin2: cheers :) that might help`
`22:59.40`	`Ritalin2`	`but transfer=yes needs threewaycalling=yes :^D`
`22:59.53`	`*** join/#asterisk MrY (n=silencer@66-7-233-146.static-ip.telepacific.net)`
`23:00.08`	`HushPe`	`legend, that's working now :)`
`23:00.08`	`Zodiacal`	`anyone know if this headset is an ear bud type? http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880043/prod4700006?prodfind=true&mftr=POLYCOM`
`23:01.06`	`HushPe`	`that's legend, even has call announce!`
`23:01.06`	`[TK]D-Fender`	`Zodiacal : Voicetube seriously sucks. Get a foam covered noise cancelling model like the H261`
`23:01.20`	`MrY`	`i have asterisk 1.2.14 installed - i see the program mpg123 running and it's taking up like 50% of cpu time.. why?`
`23:01.34`	`HushPe`	`MrY: i just had that`
`23:01.39`	`HushPe`	`kill asterisk, and restart it`
`23:01.40`	`Zodiacal`	`TKD-fender really? you have tried both?`
`23:01.51`	`ThoMe`	`kann hie rauch einer deutsch?! :_9`
`23:01.53`	`Zodiacal`	`TKD-fender i havn't tried either :)`
`23:01.54`	`MrY`	`HushPe: everytime you have that, you have to kill it?`
`23:01.56`	`HushPe`	`i suspect it's a rogue process`
`23:02.15`	`HushPe`	`no, it was just a rouge process (or 3) in my case`
`23:02.26`	`MrY`	`asterisk i stopped it.. but i still see the 2 daemons there`
`23:02.33`	`Zodiacal`	`TK-D-fender do you have a fav. headset?`
`23:02.44`	`HushPe`	`kill them manuall`
`23:02.49`	`[TK]D-Fender`	`Zodiacal : I've gone through a number of these in my call center. the H261 really wins.`
`23:02.49`	`HushPe`	`then restart asterisk`
`23:02.56`	`HushPe`	`rogue being the main word ;)`
`23:03.05`	`MrY`	`bug in asterisk?`
`23:03.06`	`MrY`	`:)`
`23:03.12`	`Ritalin2`	`ThoMe: was?`
`23:03.16`	`MrY`	`my gentoo install doesn't have 1.4 yet`
`23:03.20`	`MrY`	`i wish i can install 1.4`
`23:03.43`	`Zodiacal`	`tkD-fender looking for one that goes behind the neck`
`23:03.48`	`ThoMe`	`Ritalin2: wollt nr wissen ob hier auch einer deutsch spricht :-)`
`23:03.50`	`MrY`	`HushPe: i know mpg123, my question is why is it there all the time?`
`23:04.06`	`Ritalin2`	`ThoMe: Ich kann ein bisschen sprechen. aber nicht sehr gut`
`23:04.19`	`ThoMe`	`Ritalin2: cool, wie kommts? wo hast du es gelernt?`
`23:04.20`	`*** join/#asterisk seva (i=seva@66.90.103.12)`
`23:04.34`	`Zodiacal`	`tkd-fender like this guy? http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880043/prod4700007?prodfind=true&mftr=POLYCOM`
`23:04.38`	`Ritalin2`	`ThoMe: ich lerne es im Uni. Komme aus Amerika`
`23:04.45`	`seva`	`what are suggested fax solutions for asterisk, i want to be able to send/receive from the desktop`
`23:04.50`	`HushPe`	`MrY: i think i've worked out why`
`23:04.53`	`ThoMe`	`Ritalin2: cool. :-) udn du hast auch asterisk am laufen oder was?`
`23:05.01`	`ThoMe`	`Ritalin2: bist du fit mit asterisk?`
`23:05.03`	`HushPe`	`i did killall -HUP asterisk to reload`
`23:05.08`	`HushPe`	`and i have the rogue processes`
`23:05.14`	`[TK]D-Fender`	`Zodiacal : Ok, well keep looking, go binaural, and make sure to get the NS mic,. Voicetube catches wind too much and breathing sounds like a hurricane...`
`23:05.37`	`Zodiacal`	`TK D-fender wahts binaural off hand?`
`23:05.39`	`[TK]D-Fender`	`Zodiacal : Falls under the category of "too good for its own good".`
`23:05.56`	`Ritalin2`	`ThoMe ein wenig :) haben Sie ein Frage?`
`23:05.57`	`[TK]D-Fender`	`Zodiacal : over both ears. Lets your CSR's focus on their callers`
`23:06.16`	`Zodiacal`	`oh like ear muffs?`
`23:06.51`	`Zodiacal`	`sometimes the users will need to hear other local people`
`23:07.01`	`[TK]D-Fender`	`Zodiacal : there are differen sizes available. I personally would find anthing that ENCASED my ears very obnoxious. Shuold be secure, but not "tight"`
`23:07.12`	`MrY`	`HushPe: so no way to solve it right?`
`23:07.15`	`[TK]D-Fender`	`Zodiacal : For that... volume control (M12 amp)`
`23:07.37`	`Zodiacal`	`would a polycom 601 be able to control the volume?`
`23:07.43`	`Zodiacal`	`or would i need an amp still?`
`23:07.50`	`[TK]D-Fender`	`Zodiacal : Sure, but you still need an amp....`
`23:07.54`	`Zodiacal`	`oh`
`23:07.57`	`ThoMe`	`Ritalin2: ach.. mich wuerde interessieren ob man mit set(CALLERID(name)=bla) auch zeilenumbrueche machen kannß`
`23:07.58`	`elriah`	`On moving numbers/dids from one telco to another, anybody do this? is it painful?`
`23:08.14`	`[TK]D-Fender`	`Zodiacal : better to let the amp do its job and leave the phone on 50%`
`23:08.25`	`*** part/#asterisk seva (i=seva@66.90.103.12)`
`23:08.25`	`qdk`	`elriah: In Denmark it takes at least a month.`
`23:08.45`	`elriah`	`Any U.S. customer nightmares to report?`
`23:09.20`	`Zodiacal`	`TK D-fender ok, thank you for the info! i'll go get one and try it...`
`23:09.34`	`Ritalin2`	`ThoMe: was ist zeilenumbrueche ?`
`23:09.57`	`ThoMe`	`Ritalin2: \n <br> breaks ;)`
`23:10.05`	`rene-`	`[TK-Dfender]: Hey man how is it going`
`23:10.26`	`HushPe`	`MrY: not killalling asterisk, you're probably right it's a bug... asterisk isn't keeping track of the mpg123 processes it's launching`
`23:10.30`	`*** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net)`
`23:10.37`	`[TK]D-Fender`	`rene- : "still breathing", and about to head out the door. Martial arts tonight...`
`23:11.18`	`[TK]D-Fender`	`Zodiacal : NP. I theid a "direct connect" on, and believe me... no amp = suck. Binaural helps, and voicetube... bleh`
`23:11.30`	`Ritalin2`	`you can't have a carriage return in the CallerID string can you?`
`23:11.53`	`elriah`	`During a number transfer process, do the numbers go dark or do they just suddently start working on the new service?`
`23:11.55`	`[TK]D-Fender`	`Zodiacal : I USED to thing it was nice, until the clarity gave way to "just too damned sensitive". This is clearly MCI's headset of choice...`
`23:12.18`	`ThoMe`	`Ritalin2: hm? ich moechte returns mit in der caller-id (name) bps "Thomas\n1234567"`
`23:12.39`	`Zodiacal`	`TKd-fender have you tried wireless ones?`
`23:12.52`	`Ritalin2`	`ThoMe: ich weiss nicht aber ich frage jemand auf englich :)`
`23:13.05`	`ThoMe`	`Ritalin2: hihi, danke :-)`
`23:13.18`	`[TK]D-Fender`	`Zodiacal : I've tried Plantronics wireless lifter one's yeah. TOTALLY wasted time & money.`
`23:13.44`	`[TK]D-Fender`	`Zodiacal : You can't DO anything but lift the handset to answer (for which you'll have to mod the phone to even work).`
`23:13.58`	`[TK]D-Fender`	`Zodiacal : If yuo want wireless, get an ATA + cordless phone.`
`23:14.06`	`UVSoft`	`hi there! there's a question, i've got one FXO and one FXS devices. if the first phone (phone1, connected to the FXO device through TLS) hangs up, i hear from the other one (phone2, connected to the FXS devices) short dial tones, than looooong dial tone, so i can dial next number (just like it should be), _but_ the FXO device itself doesn't hangs up, so when i dial a number it goes right through the FXO to the TLS without my dialplan at all!!`
`23:14.06`	`UVSoft`	`! so the TLS tries to dial the second number (and what if i want to dial to a SIP user or something). it's wrong! i want to be able to dial again without hanging up and picking up... and with my dial plan. does anybody know how to do it?`
`23:14.06`	`[TK]D-Fender`	`ok, I've got to run. Back in a few hours.`
`23:14.13`	`rene-`	`cya`
`23:14.51`	`Zodiacal`	`cya and thanks again`
`23:15.55`	`Ritalin2`	`ThoMe: ich dinke es nicht moeglich`
`23:16.22`	`ThoMe`	`Ritalin2: nagut. ok danke :_9`
`23:17.16`	`Ritalin2`	`ThoMe: ich hoffe mein Detusch ist nicht zu schlect`
`23:17.56`	`rudholm`	`Strom_C: that sounds fair.`
`23:18.18`	`Strom_C`	`:)`
`23:18.30`	`ThoMe`	`Ritalin2: nein, im gegenteil, sehr gut. ein paar rechtschreibfehler, aber jeder deutsche versteht dich einwandfrei. :-)`
`23:18.39`	`ThoMe`	`Ritalin2: und besser ist dein deutsch zu meinem englisch alle male :-)`
`23:19.26`	`Ritalin2`	`jaja`
`23:20.15`	`ThoMe`	`Ritalin2: kennst du eigentlich ne gute GUI fuer asteriskß`
`23:20.30`	`ThoMe`	`Ritalin2: hab mir jetzt ersmtal das geholt: http://www.pbx-manager.de/ ist auch ganz ok...`
`23:20.32`	`Ritalin2`	`ThoMe: FreePBX`
`23:20.57`	`*** join/#asterisk Skarmeth (n=Skarmeth@201009012089.user.veloxzone.com.br)`
`23:21.14`	`Ritalin2`	`ThoMe: also... www.trixbox.org`
`23:23.15`	`EmleyMoor`	`Anyone got a number I can reach over SIPBroker?`
`23:23.51`	`ThoMe`	`Ritalin2: siehe query.`
`23:24.09`	`cekc`	`emley: you in AU?`
`23:24.16`	`EmleyMoor`	`UK`
`23:24.48`	`EmleyMoor`	`If I was from AU, I'd have a hat with corks dangling from it <g>`
`23:25.05`	`cekc`	`if you were in AU you could call +61-2-8214-6640`
`23:25.29`	`EmleyMoor`	`I'm more after an ID on a VoIP provider...`
`23:26.04`	`EmleyMoor`	`Australia is inexpensive on VoIPtalk :-)`
`23:27.03`	`matt_`	`does anybody know of a service that will allow me to route sip calls to the tesco voip network?`
`23:27.25`	`matt_`	`i dont want to buy a stupid phone just to get an account`
`23:28.42`	`*** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk)`
`23:30.20`	`matt_`	`EmleyMoor, i was just reading into that, theres an intresting story here .. http://blog.tmcnet.com/blog/tom-keating/skype/sip-to-skype-gateway-breaks-skypes-great-wall-of-voip.asp`
`23:31.58`	`matt_`	`i can easily see it becomming a big mess tho, like i would have to have multiple numbers just some people on different networks can call me`
`23:32.26`	`EmleyMoor`	`Indeed`
`23:32.57`	`*** join/#asterisk malverian (n=malveria@gentoo/developer/malverian)`
`23:33.01`	`matt_`	`i really hope there will be a centeral enum database where people can get free numbers and add voip addresses and that number will work on every network with voip support`
`23:33.17`	`EmleyMoor`	`# We shall overcome...`
`23:33.28`	`EmleyMoor`	`# We shall overcome...`
`23:33.38`	`EmleyMoor`	`# We shall overcome some day #`
`23:34.22`	`EmleyMoor`	`I ought to be going to bed soon`
`23:34.43`	`matt_`	`do you know what the tesco service is like ?`
`23:34.52`	`matt_`	`somebody was saying the quality is rubbish`
`23:36.42`	`*** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`23:36.43`	`*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)`
`23:38.23`	`matt_`	`humm looks like tesco voip uses iax`
`23:38.31`	`matt_`	`wonder if they accept guest connections`
`23:38.34`	`matt_`	`bet they dont`
`23:38.41`	`marv[work]`	`in 1.4, how do you get make to give you the raw output, instead of the pretty [useless] output it defaults to`
`23:40.43`	`*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)`
`23:40.57`	`syzygyBSD`	`wow, what a pain`
`23:41.31`	`*** part/#asterisk dendrite (n=ph@dsl092-150-203.wdc2.dsl.speakeasy.net)`
`23:41.33`	`syzygyBSD`	`of all the things to go wrong on a server, the onboard raid dies`
`23:41.59`	`marv[work]`	`why are you using onboard raid?`
`23:42.04`	`marv[work]`	`wait, you're not anymore`
`23:42.10`	`*** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net)`
`23:42.18`	`syzygyBSD`	`lol, sure I am, got a replacement MB`
`23:42.35`	`marv[work]`	`ah, ok`
`23:42.35`	`syzygyBSD`	`didn't realize it was the onboard raid till everything was working again and I could track it down`
`23:42.45`	`syzygyBSD`	`afterwards the entire mb died...`
`23:42.52`	`*** join/#asterisk dendrite (n=ph@dsl092-150-203.wdc2.dsl.speakeasy.net)`
`23:42.58`	`marv[work]`	`wait, the one with the bad raid, or the new one?`
`23:43.16`	`syzygyBSD`	`sorry, not onboard raid, onboard scsi controller`
`23:43.29`	`syzygyBSD`	`raid was done via software...`
`23:43.45`	`marv[work]`	`oh, that's a slight difference`
`23:43.49`	`syzygyBSD`	`the mb with the bad scsi controller died`
`23:44.05`	`syzygyBSD`	`ya, but we only have raided scsi drives...`
`23:44.31`	`syzygyBSD`	`I can't wait till I get my new SATA controller.. will be fun times`
`23:44.57`	`syzygyBSD`	`trying to fit 50 HD's into 1 4U box`
`23:45.42`	`Nugget`	`plus you'll be able to use it to cook pop tarts.`
`23:46.13`	`syzygyBSD`	`indeed`
`23:46.22`	`syzygyBSD`	`low access HD's though`
`23:46.25`	`*** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com)`
`23:47.04`	`syzygyBSD`	`just make sure you clean the crumb tray`
`23:47.35`	`*** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner)`
`23:48.47`	`Ritalin2`	`whats the exten for dropping X number of digits?`
`23:49.38`	`*** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk)`
`23:49.44`	`matt_`	`humm theres somebody saying here that they registered 2 tesco accounts is that possiable ?`
`23:50.02`	`matt_`	`does anybody know if you can register at a site`
`23:52.09`	`*** join/#asterisk NeonLevel (n=NeonLeve@189.169.21.36)`
`23:52.15`	`lenne_dk`	`cat the astdb be accessed from php or perl?`
`23:52.52`	`*** join/#asterisk smackus (n=ckwall@63.149.122.93)`
`23:54.10`	`smackus`	`i am having an iax connection go up and down over and over. i set qualify=yes on each side of the account. When the connection is up, it works great, but then it drops. I have looked at udp dump... nothing. I have tried to make heads or tails of the debug output. I dont see anything the jumps out at me... anything else I can try?`
`23:54.40`	`matt_`	`smackus, have you tried quality=no ?`
`23:55.05`	`smackus`	`quality=no? I kinda want it to be quality`
`23:55.09`	`smackus`	`did you mean qualify?`
`23:55.14`	`Corydon-w`	`smackus: qualifysmoothing=yes`
`23:55.24`	`smackus`	`is that in 1.2.14?`
`23:55.28`	`Corydon-w`	`Yes`
`23:55.30`	`smackus`	`ok`
`23:55.32`	`smackus`	`cool`
`23:55.49`	`smackus`	`with that can i use qualify=yes? so i can see that status?`
`23:56.02`	`Corydon-w`	`You have to`
`23:56.06`	`smackus`	`ok...`
`23:56.11`	`Corydon-w`	`It only means something when qualify=yes`
`23:56.13`	`ThoMe`	`Corydon-w: what is ** ? _XX ?`
`23:56.19`	`ThoMe`	`i have`
`23:56.19`	`ThoMe`	`exten => _XX,1,Dial(SIP/${EXTEN},30)`
`23:56.28`	`ThoMe`	`but not 0051 << > **51...?`
`23:56.29`	`ThoMe`	`how?`
`23:56.47`	`Corydon-w`	`ThoMe: two digits`
`23:57.03`	`Corydon-w`	`* is not a digit`
`23:57.05`	`ThoMe`	`Corydon-w: and if i want "**" ?`
`23:57.14`	`Corydon-w`	`_**XX`
`23:57.20`	`De_Mon`	`how unique are priority labels? context, or extension?`
`23:57.22`	`smackus`	`and would i have experienced a call not going through over the iax because qualify=yes was set and its time exceeded? or would that have happened regardless of the qualify = value?`
`23:57.43`	`Corydon-w`	`De_Mon: extension`
`23:57.50`	`smackus`	`i mean would qualify=yes made it so the call did not go through since it found it was not responsive?`
`23:57.53`	`Corydon-w`	`De_Mon: actually, both`
`23:58.18`	`ThoMe`	`Corydon-w: i have now: exten => _**XX,1,Dial(SIP/${EXTEN},30)`
`23:58.24`	`ThoMe`	`but: Jan 26 00:57:58 NOTICE[12612]: chan_iax2.c:7331 socket_read: Rejected connect attempt from 194.231.22.135, request '**50@from-inside' does not exist`
`23:58.29`	`Ritalin2`	`Corydon: how do you truncate the first so many numbers off an extension?`
`23:58.34`	`De_Mon`	`both? this is an either or type question`
`23:58.35`	`smackus`	`well, i gotta run. i will play with this more and come back for help if needed. thanks`
`23:58.38`	`Corydon-w`	`smackus: qualifysmoothing requires two bad responses in a row for the peer to be considered unavailable`
`23:58.38`	`*** part/#asterisk smackus (n=ckwall@63.149.122.93)`
`23:59.04`	`De_Mon`	`either I can add exten1,n(label) and exten2,n(label) or it says that label already exists.`
`23:59.13`	`Corydon-w`	`De_Mon: so same extension, different context, means the label is different too`
`23:59.14`	`De_Mon`	`guess I should just test`
`23:59.25`	`Corydon-w`	`De_Mon: yes`
`23:59.50`	`Corydon-w`	`or same context, different extension, the label is still different`
`23:59.56`	`De_Mon`	`same extension in different contexts is still a yeah`

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